voicemail and autoattendant 408 timeout

[m***@mattkeys.net]

Hello,

I'm having the same problem as this user : http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331

I can make calls from extension to extension, however if I let it go to voicemail the system returns a "408 Timeout" to the phone. Calls directly to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000 will return "408 Timeout" on the display screen. The CDRs also show the 408 and a failed status. In sipXproxy.log, I see (I've sensored the IP with xxx.xxx.xxx.xxx) :

"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction"
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write 22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore OsSocket::write() returned -1, errno = 32"
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction"

Thanks,
Matt

[m***@mattkeys.net]

Whoops, left the hostname there! Oh well...

I also found a bug report about this here : http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel . Per the request of the developer on that page I've attached a INFO level log of it.


From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of ***@mattkeys.net
Sent: Saturday, June 16, 2012 9:39 AM
To: sipx-***@list.sipfoundry.org
Subject: [sipx-users] voicemail and autoattendant 408 timeout

Hello,

I'm having the same problem as this user : http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331

I can make calls from extension to extension, however if I let it go to voicemail the system returns a "408 Timeout" to the phone. Calls directly to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000 will return "408 Timeout" on the display screen. The CDRs also show the 408 and a failed status. In sipXproxy.log, I see (I've sensored the IP with xxx.xxx.xxx.xxx) :

"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction"
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write 22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore OsSocket::write() returned -1, errno = 32"
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction"

Thanks,
Matt

[Tony Graziano]

You are not really providing enough information...

Is the phone local or remote? If it is remote, did you change any of the
phone parameters manually (I ask because the SRV records are missing for
the domain name and that will always cause issues)...

It would really help if you explained the call flow and provided a
siptrace. snippets of logs tell not much...

http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer

Whoops, left the hostname there! Oh well… ****
** **
http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel. Per the request of the developer on that page I've attached a INFO level
log of it.****
** **
** **
*Sent:* Saturday, June 16, 2012 9:39 AM
*Subject:* [sipx-users] voicemail and autoattendant 408 timeout****
** **
Hello,****
** **
http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331
****
** **
I can make calls from extension to extension, however if I let it go to
voicemail the system returns a "408 Timeout" to the phone. Calls directly
to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000
will return "408 Timeout" on the display screen. The CDRs also show the 408
and a failed status. In sipXproxy.log, I see (I've sensored the IP with
xxx.xxx.xxx.xxx) :****
** **
"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"****
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write
22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"**
**
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore
OsSocket::write() returned -1, errno = 32"****
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"****
** **
Thanks,****
Matt****
** **
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

[m***@mattkeys.net]

Hi Tony, thanks for the quick response. There are phones in both locations. I'm testing using a Grandstream GXV3000 remotely from home but there are two Polycom 321s also registered on the premises. SRV records are there, you've just got to look in the right spot. The grandstream here is pointed to the DNS there, the Polycoms on site use the sipXecs server for DHCP/DNS and gateway :

~]$ dig @70.88.18.153 SRV _sip._tcp.hmherbs.com

; <<>> DiG 9.7.3-P3-RedHat-9.7.3-8.P3.el6_2.3 <<>> @70.88.18.153 SRV _sip._tcp.hmherbs.com
; (1 server found)
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1

;; QUESTION SECTION:
;_sip._tcp.hmherbs.com. IN SRV

;; ANSWER SECTION:
_sip._tcp.hmherbs.com. 1800 IN SRV 1 0 5060 sipx.hmherbs.com.

;; AUTHORITY SECTION:
hmherbs.com. 1800 IN NS sipx.hmherbs.com.

;; ADDITIONAL SECTION:
sipx.hmherbs.com. 1800 IN A 70.88.18.153

;; Query time: 48 msec
;; SERVER: 70.88.18.153#53(70.88.18.153)
;; WHEN: Sat Jun 16 14:52:03 2012
;; MSG SIZE rcvd: 105

I've attached the siptrace of the call to 101 I attached in the earlier logs.

Thanks again,
Matt

From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Saturday, June 16, 2012 11:24 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

You are not really providing enough information...

Is the phone local or remote? If it is remote, did you change any of the phone parameters manually (I ask because the SRV records are missing for the domain name and that will always cause issues)...

It would really help if you explained the call flow and provided a siptrace. snippets of logs tell not much...

http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
On Sat, Jun 16, 2012 at 10:37 AM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Whoops, left the hostname there! Oh well...

I also found a bug report about this here : http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel . Per the request of the developer on that page I've attached a INFO level log of it.


From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of ***@mattkeys.net<mailto:***@mattkeys.net>
Sent: Saturday, June 16, 2012 9:39 AM
To: sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
Subject: [sipx-users] voicemail and autoattendant 408 timeout

Hello,

I'm having the same problem as this user : http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331

I can make calls from extension to extension, however if I let it go to voicemail the system returns a "408 Timeout" to the phone. Calls directly to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000 will return "408 Timeout" on the display screen. The CDRs also show the 408 and a failed status. In sipXproxy.log, I see (I've sensored the IP with xxx.xxx.xxx.xxx) :

"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction"
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write 22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore OsSocket::write() returned -1, errno = 32"
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction"

Thanks,
Matt


_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

[Tony Graziano]

its a grandstream having the problem or both?

Can you tell me how the phones were programmed? I think I see the hostname
being called "instead of" the domain name. Why would that be?

Did you properly populate the intranet subnets?

Post by m***@mattkeys.net
Hi Tony, thanks for the quick response. There are phones in both
locations. I'm testing using a Grandstream GXV3000 remotely from home but
there are two Polycom 321s also registered on the premises. SRV records are
there, you've just got to look in the right spot. The grandstream here is
pointed to the DNS there, the Polycoms on site use the sipXecs server for
DHCP/DNS and gateway :****
** **
** **
_sip._tcp.hmherbs.com****
; (1 server found)****
;; global options: +cmd****
;; Got answer:****
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630****
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1***
*
** **
;; QUESTION SECTION:****
;_sip._tcp.hmherbs.com. IN SRV****
** **
;; ANSWER SECTION:****
_sip._tcp.hmherbs.com. 1800 IN SRV 1 0 5060 sipx.hmherbs.com.
****
** **
;; AUTHORITY SECTION:****
hmherbs.com. 1800 IN NS sipx.hmherbs.com.****
** **
;; ADDITIONAL SECTION:****
sipx.hmherbs.com. 1800 IN A 70.88.18.153****
** **
;; Query time: 48 msec****
;; SERVER: 70.88.18.153#53(70.88.18.153)****
;; WHEN: Sat Jun 16 14:52:03 2012****
;; MSG SIZE rcvd: 105****
** **
I've attached the siptrace of the call to 101 I attached in the earlier
logs.****
** **
Thanks again,****
Matt****
** **
*Sent:* Saturday, June 16, 2012 11:24 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
** **
You are not really providing enough information...****
** **
Is the phone local or remote? If it is remote, did you change any of the
phone parameters manually (I ask because the SRV records are missing for
the domain name and that will always cause issues)...****
** **
It would really help if you explained the call flow and provided a
siptrace. snippets of logs tell not much...****
** **
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
****
wrote:****
Whoops, left the hostname there! Oh well… ****
****
http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel. Per the request of the developer on that page I've attached a INFO level
log of it.****
****
****
*Sent:* Saturday, June 16, 2012 9:39 AM
*Subject:* [sipx-users] voicemail and autoattendant 408 timeout****
****
Hello,****
****
http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331
****
****
I can make calls from extension to extension, however if I let it go to
voicemail the system returns a "408 Timeout" to the phone. Calls directly
to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000
will return "408 Timeout" on the display screen. The CDRs also show the 408
and a failed status. In sipXproxy.log, I see (I've sensored the IP with
xxx.xxx.xxx.xxx) :****
****
"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"****
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write
22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"**
**
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore
OsSocket::write() returned -1, errno = 32"****
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"****
****
Thanks,****
Matt****
****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
** **
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
** **
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
** **
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

[m***@mattkeys.net]

It's both sides with this problem. The polycom 321s behind the server are pulling programming/provisioning from sipxecs (dhcp and tftp). The grandstream I have here at home I've configured manually. Again, I can successfully make calls to/from all extensions, audio is working bidirectionally, and I see registrations for all of them. The hostname (sipx.hmherbs.com) will probably be a full subdomain soon but at the moment I've just got it set on a A record and the clients pointing to sipxecs for DNS. Under System -> Domain, the domain name is set to "sipx.hmherbs.com", with an alias for voice.hmherbs.com and hmherbs.com set, depending on what the customer wants to use when we're finished. Under System -> Internet Calling, I've got 192.168.1.0/24, 192.168.2.0/24, and 10.1.10.0/24 as Intranet subnets and *.hmherbs.com as Intranet domains.

I'm thinking it has to do with the " OsSocket::write() returned -1, errno = 32 " and broken pipe messages... but I didn't know for sure what's causing it.

Thanks,
Matt

From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Saturday, June 16, 2012 5:51 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

its a grandstream having the problem or both?

Can you tell me how the phones were programmed? I think I see the hostname being called "instead of" the domain name. Why would that be?

Did you properly populate the intranet subnets?
On Sat, Jun 16, 2012 at 3:07 PM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Hi Tony, thanks for the quick response. There are phones in both locations. I'm testing using a Grandstream GXV3000 remotely from home but there are two Polycom 321s also registered on the premises. SRV records are there, you've just got to look in the right spot. The grandstream here is pointed to the DNS there, the Polycoms on site use the sipXecs server for DHCP/DNS and gateway :

~]$ dig @70.88.18.153<http://70.88.18.153> SRV _sip._tcp.hmherbs.com<http://tcp.hmherbs.com>

; <<>> DiG 9.7.3-P3-RedHat-9.7.3-8.P3.el6_2.3 <<>> @70.88.18.153<http://70.88.18.153> SRV _sip._tcp.hmherbs.com<http://tcp.hmherbs.com>
; (1 server found)
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1

;; QUESTION SECTION:
;_sip._tcp.hmherbs.com<http://tcp.hmherbs.com>. IN SRV

;; ANSWER SECTION:
_sip._tcp.hmherbs.com<http://tcp.hmherbs.com>. 1800 IN SRV 1 0 5060 sipx.hmherbs.com<http://sipx.hmherbs.com>.

;; AUTHORITY SECTION:
hmherbs.com<http://hmherbs.com>. 1800 IN NS sipx.hmherbs.com<http://sipx.hmherbs.com>.

;; ADDITIONAL SECTION:
sipx.hmherbs.com<http://sipx.hmherbs.com>. 1800 IN A 70.88.18.153

;; Query time: 48 msec
;; SERVER: 70.88.18.153#53(70.88.18.153)
;; WHEN: Sat Jun 16 14:52:03 2012
;; MSG SIZE rcvd: 105

I've attached the siptrace of the call to 101 I attached in the earlier logs.

Thanks again,
Matt

From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Saturday, June 16, 2012 11:24 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

You are not really providing enough information...

Is the phone local or remote? If it is remote, did you change any of the phone parameters manually (I ask because the SRV records are missing for the domain name and that will always cause issues)...

It would really help if you explained the call flow and provided a siptrace. snippets of logs tell not much...

http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
On Sat, Jun 16, 2012 at 10:37 AM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Whoops, left the hostname there! Oh well...

I also found a bug report about this here : http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel . Per the request of the developer on that page I've attached a INFO level log of it.


From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of ***@mattkeys.net<mailto:***@mattkeys.net>
Sent: Saturday, June 16, 2012 9:39 AM
To: sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
Subject: [sipx-users] voicemail and autoattendant 408 timeout

Hello,

I'm having the same problem as this user : http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331

I can make calls from extension to extension, however if I let it go to voicemail the system returns a "408 Timeout" to the phone. Calls directly to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000 will return "408 Timeout" on the display screen. The CDRs also show the 408 and a failed status. In sipXproxy.log, I see (I've sensored the IP with xxx.xxx.xxx.xxx) :

"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction"
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write 22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore OsSocket::write() returned -1, errno = 32"
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction"

Thanks,
Matt


_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

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[Tony Graziano]

What is the output of
sipxproc --state
Does this happen on every call to the voicemail? When you set the system up
did you send the server its profile?

If it were me I'd kick the grand streams to the curb. Since you have them
you might be better off to remove the other codecs except g711 from them. I
still think something is wring with the domain aspect of it.

Ensure you have a domain alias of the "hostname" and restart services as
prompted. When you decide to do a sub domain in production, wipe and
rebuild the system from bare metal, don't try to reconfigure it.

Also, check top to make sure the system is not using swap.

Post by m***@mattkeys.net
It's both sides with this problem. The polycom 321s behind the server are
pulling programming/provisioning from sipxecs (dhcp and tftp). The
grandstream I have here at home I've configured manually. Again, I can
successfully make calls to/from all extensions, audio is working
bidirectionally, and I see registrations for all of them. The hostname (
sipx.hmherbs.com) will probably be a full subdomain soon but at the
moment I've just got it set on a A record and the clients pointing to
sipxecs for DNS. Under System -> Domain, the domain name is set to "
sipx.hmherbs.com", with an alias for voice.hmherbs.com and hmherbs.comset, depending on what the customer wants to use when we're finished. Under
System -> Internet Calling, I've got 192.168.1.0/24, 192.168.2.0/24, and
10.1.10.0/24 as Intranet subnets and *.hmherbs.com as Intranet domains.***
*
** **
I'm thinking it has to do with the " OsSocket::write() returned -1, errno
= 32 " and broken pipe messages… but I didn't know for sure what's causing
it.****
** **
Thanks,****
Matt****
** **
*Sent:* Saturday, June 16, 2012 5:51 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
** **
its a grandstream having the problem or both?****
** **
Can you tell me how the phones were programmed? I think I see the hostname
being called "instead of" the domain name. Why would that be? ****
** **
Did you properly populate the intranet subnets?****
wrote:****
Hi Tony, thanks for the quick response. There are phones in both
locations. I'm testing using a Grandstream GXV3000 remotely from home but
there are two Polycom 321s also registered on the premises. SRV records are
there, you've just got to look in the right spot. The grandstream here is
pointed to the DNS there, the Polycoms on site use the sipXecs server for
DHCP/DNS and gateway :****
****
****
_sip._tcp.hmherbs.com****
; (1 server found)****
;; global options: +cmd****
;; Got answer:****
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630****
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1***
*
****
;; QUESTION SECTION:****
;_sip._tcp.hmherbs.com. IN SRV****
****
;; ANSWER SECTION:****
_sip._tcp.hmherbs.com. 1800 IN SRV 1 0 5060 sipx.hmherbs.com.
****
****
;; AUTHORITY SECTION:****
hmherbs.com. 1800 IN NS sipx.hmherbs.com.****
****
;; ADDITIONAL SECTION:****
sipx.hmherbs.com. 1800 IN A 70.88.18.153****
****
;; Query time: 48 msec****
;; SERVER: 70.88.18.153#53(70.88.18.153)****
;; WHEN: Sat Jun 16 14:52:03 2012****
;; MSG SIZE rcvd: 105****
****
I've attached the siptrace of the call to 101 I attached in the earlier
logs.****
****
Thanks again,****
Matt****
****
*Sent:* Saturday, June 16, 2012 11:24 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
You are not really providing enough information...****
****
Is the phone local or remote? If it is remote, did you change any of the
phone parameters manually (I ask because the SRV records are missing for
the domain name and that will always cause issues)...****
****
It would really help if you explained the call flow and provided a
siptrace. snippets of logs tell not much...****
****
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
****
wrote:****
Whoops, left the hostname there! Oh well… ****
****
http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel. Per the request of the developer on that page I've attached a INFO level
log of it.****
****
****
*Sent:* Saturday, June 16, 2012 9:39 AM
*Subject:* [sipx-users] voicemail and autoattendant 408 timeout****
****
Hello,****
****
http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331
****
****
I can make calls from extension to extension, however if I let it go to
voicemail the system returns a "408 Timeout" to the phone. Calls directly
to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000
will return "408 Timeout" on the display screen. The CDRs also show the 408
and a failed status. In sipXproxy.log, I see (I've sensored the IP with
xxx.xxx.xxx.xxx) :****
****
"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"****
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write
22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"**
**
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore
OsSocket::write() returned -1, errno = 32"****
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"****
****
Thanks,****
Matt****
****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
****
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
****
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
****
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
** **
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
** **
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
** **
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/

[m***@mattkeys.net]

Yes it's every call to voicemail (101), operator (0), or if I call an extension and let it go to voicemail. As you've guessed already this is the initial dev/test installation, so if it would help our troubleshooting I'd be happy to set up an extension for you. The SIP trunks haven't been terminated to it yet so it could only be strictly extension to extension dialing. I couldn't remember if I had sent the server profiles so I've just done that, added intranet domain alias *.sipx.hmherbs.com, *.voice.hmherbs.com, and restarted the box. Here's the output you asked for :

[***@sipx sipxpbx]# sipxproc --state
{"FreeSWITCH"=>"ConfigurationTestFailed",
"ConfigServer"=>"Running",
"SipXbridge"=>"Running",
"ConfigAgent"=>"Disabled",
"sipXprovision"=>"Running",
"SipXopenfire"=>"Running",
"PresenceServer"=>"Disabled",
"SipXrelay"=>"Running",
"sipXimbot"=>"Running",
"sipXmrtg"=>"Running",
"SIPRegistrar"=>"Running",
"CallResolver-Agent"=>"Disabled",
"ACDServer"=>"Disabled",
"sipXivr"=>"Running",
"PageServer"=>"Running",
"sipXacccode"=>"Running",
"ParkServer"=>"Running",
"ResourceListServer"=>"Running",
"SharedAppearanceAgent"=>"Running",
"SIPStatus"=>"Running",
"sipXrecording"=>"Running",
"CallResolver"=>"Running",
"SipXrest"=>"Running",
"SIPXProxy"=>"Running"}

From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Sunday, June 17, 2012 6:12 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout


What is the output of
sipxproc --state
Does this happen on every call to the voicemail? When you set the system up did you send the server its profile?

If it were me I'd kick the grand streams to the curb. Since you have them you might be better off to remove the other codecs except g711 from them. I still think something is wring with the domain aspect of it.

Ensure you have a domain alias of the "hostname" and restart services as prompted. When you decide to do a sub domain in production, wipe and rebuild the system from bare metal, don't try to reconfigure it.

Also, check top to make sure the system is not using swap.
On Jun 16, 2012 9:22 PM, "***@mattkeys.net<mailto:***@mattkeys.net>" <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
It's both sides with this problem. The polycom 321s behind the server are pulling programming/provisioning from sipxecs (dhcp and tftp). The grandstream I have here at home I've configured manually. Again, I can successfully make calls to/from all extensions, audio is working bidirectionally, and I see registrations for all of them. The hostname (sipx.hmherbs.com<http://sipx.hmherbs.com>) will probably be a full subdomain soon but at the moment I've just got it set on a A record and the clients pointing to sipxecs for DNS. Under System -> Domain, the domain name is set to "sipx.hmherbs.com<http://sipx.hmherbs.com>", with an alias for voice.hmherbs.com<http://voice.hmherbs.com> and hmherbs.com<http://hmherbs.com> set, depending on what the customer wants to use when we're finished. Under System -> Internet Calling, I've got 192.168.1.0/24<http://192.168.1.0/24>, 192.168.2.0/24<http://192.168.2.0/24>, and 10.1.10.0/24<http://10.1.10.0/24> as Intranet subnets and *.hmherbs.com<http://hmherbs.com> as Intranet domains.

I'm thinking it has to do with the " OsSocket::write() returned -1, errno = 32 " and broken pipe messages... but I didn't know for sure what's causing it.

Thanks,
Matt

From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Saturday, June 16, 2012 5:51 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

its a grandstream having the problem or both?

Can you tell me how the phones were programmed? I think I see the hostname being called "instead of" the domain name. Why would that be?

Did you properly populate the intranet subnets?
On Sat, Jun 16, 2012 at 3:07 PM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Hi Tony, thanks for the quick response. There are phones in both locations. I'm testing using a Grandstream GXV3000 remotely from home but there are two Polycom 321s also registered on the premises. SRV records are there, you've just got to look in the right spot. The grandstream here is pointed to the DNS there, the Polycoms on site use the sipXecs server for DHCP/DNS and gateway :

~]$ dig @70.88.18.153<http://70.88.18.153> SRV _sip._tcp.hmherbs.com<http://tcp.hmherbs.com>

; <<>> DiG 9.7.3-P3-RedHat-9.7.3-8.P3.el6_2.3 <<>> @70.88.18.153<http://70.88.18.153> SRV _sip._tcp.hmherbs.com<http://tcp.hmherbs.com>
; (1 server found)
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1

;; QUESTION SECTION:
;_sip._tcp.hmherbs.com<http://tcp.hmherbs.com>. IN SRV

;; ANSWER SECTION:
_sip._tcp.hmherbs.com<http://tcp.hmherbs.com>. 1800 IN SRV 1 0 5060 sipx.hmherbs.com<http://sipx.hmherbs.com>.

;; AUTHORITY SECTION:
hmherbs.com<http://hmherbs.com>. 1800 IN NS sipx.hmherbs.com<http://sipx.hmherbs.com>.

;; ADDITIONAL SECTION:
sipx.hmherbs.com<http://sipx.hmherbs.com>. 1800 IN A 70.88.18.153

;; Query time: 48 msec
;; SERVER: 70.88.18.153#53(70.88.18.153)
;; WHEN: Sat Jun 16 14:52:03 2012
;; MSG SIZE rcvd: 105

I've attached the siptrace of the call to 101 I attached in the earlier logs.

Thanks again,
Matt

From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Saturday, June 16, 2012 11:24 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

You are not really providing enough information...

Is the phone local or remote? If it is remote, did you change any of the phone parameters manually (I ask because the SRV records are missing for the domain name and that will always cause issues)...

It would really help if you explained the call flow and provided a siptrace. snippets of logs tell not much...

http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
On Sat, Jun 16, 2012 at 10:37 AM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Whoops, left the hostname there! Oh well...

I also found a bug report about this here : http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel . Per the request of the developer on that page I've attached a INFO level log of it.


From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of ***@mattkeys.net<mailto:***@mattkeys.net>
Sent: Saturday, June 16, 2012 9:39 AM
To: sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
Subject: [sipx-users] voicemail and autoattendant 408 timeout

Hello,

I'm having the same problem as this user : http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331

I can make calls from extension to extension, however if I let it go to voicemail the system returns a "408 Timeout" to the phone. Calls directly to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000 will return "408 Timeout" on the display screen. The CDRs also show the 408 and a failed status. In sipXproxy.log, I see (I've sensored the IP with xxx.xxx.xxx.xxx) :

"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction"
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write 22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore OsSocket::write() returned -1, errno = 32"
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction"

Thanks,
Matt


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List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

_______________________________________________
sipx-users mailing list
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List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

[Tony Graziano]

Your freeswitch connection test failed. Freeswicth s the media engine for
voicemail and auto attendant. Put the log level to debug for media server
and restart it and then place a failed call and look at the freswitch logs
for why!

Post by m***@mattkeys.net
Yes it's every call to voicemail (101), operator (0), or if I call an
extension and let it go to voicemail. As you've guessed already this is the
initial dev/test installation, so if it would help our troubleshooting I'd
be happy to set up an extension for you. The SIP trunks haven't been
terminated to it yet so it could only be strictly extension to extension
dialing. I couldn't remember if I had sent the server profiles so I've just
done that, added intranet domain alias *.sipx.hmherbs.com, *.
voice.hmherbs.com, and restarted the box. Here's the output you asked for
:****
** **
{"FreeSWITCH"=>"ConfigurationTestFailed",****
"ConfigServer"=>"Running",****
"SipXbridge"=>"Running",****
"ConfigAgent"=>"Disabled",****
"sipXprovision"=>"Running",****
"SipXopenfire"=>"Running",****
"PresenceServer"=>"Disabled",****
"SipXrelay"=>"Running",****
"sipXimbot"=>"Running",****
"sipXmrtg"=>"Running",****
"SIPRegistrar"=>"Running",****
"CallResolver-Agent"=>"Disabled",****
"ACDServer"=>"Disabled",****
"sipXivr"=>"Running",****
"PageServer"=>"Running",****
"sipXacccode"=>"Running",****
"ParkServer"=>"Running",****
"ResourceListServer"=>"Running",****
"SharedAppearanceAgent"=>"Running",****
"SIPStatus"=>"Running",****
"sipXrecording"=>"Running",****
"CallResolver"=>"Running",****
"SipXrest"=>"Running",****
"SIPXProxy"=>"Running"}****
** **
*Sent:* Sunday, June 17, 2012 6:12 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
** **
What is the output of
sipxproc --state
Does this happen on every call to the voicemail? When you set the system
up did you send the server its profile?****
If it were me I'd kick the grand streams to the curb. Since you have them
you might be better off to remove the other codecs except g711 from them. I
still think something is wring with the domain aspect of it.****
Ensure you have a domain alias of the "hostname" and restart services as
prompted. When you decide to do a sub domain in production, wipe and
rebuild the system from bare metal, don't try to reconfigure it.****
Also, check top to make sure the system is not using swap.****
*
It's both sides with this problem. The polycom 321s behind the server are
pulling programming/provisioning from sipxecs (dhcp and tftp). The
grandstream I have here at home I've configured manually. Again, I can
successfully make calls to/from all extensions, audio is working
bidirectionally, and I see registrations for all of them. The hostname (
sipx.hmherbs.com) will probably be a full subdomain soon but at the
moment I've just got it set on a A record and the clients pointing to
sipxecs for DNS. Under System -> Domain, the domain name is set to "
sipx.hmherbs.com", with an alias for voice.hmherbs.com and hmherbs.comset, depending on what the customer wants to use when we're finished. Under
System -> Internet Calling, I've got 192.168.1.0/24, 192.168.2.0/24, and
10.1.10.0/24 as Intranet subnets and *.hmherbs.com as Intranet domains.***
*
****
I'm thinking it has to do with the " OsSocket::write() returned -1, errno
= 32 " and broken pipe messages… but I didn't know for sure what's causing
it.****
****
Thanks,****
Matt****
****
*Sent:* Saturday, June 16, 2012 5:51 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
its a grandstream having the problem or both?****
****
Can you tell me how the phones were programmed? I think I see the hostname
being called "instead of" the domain name. Why would that be? ****
****
Did you properly populate the intranet subnets?****
wrote:****
Hi Tony, thanks for the quick response. There are phones in both
locations. I'm testing using a Grandstream GXV3000 remotely from home but
there are two Polycom 321s also registered on the premises. SRV records are
there, you've just got to look in the right spot. The grandstream here is
pointed to the DNS there, the Polycoms on site use the sipXecs server for
DHCP/DNS and gateway :****
****
****
_sip._tcp.hmherbs.com****
; (1 server found)****
;; global options: +cmd****
;; Got answer:****
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630****
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1***
*
****
;; QUESTION SECTION:****
;_sip._tcp.hmherbs.com. IN SRV****
****
;; ANSWER SECTION:****
_sip._tcp.hmherbs.com. 1800 IN SRV 1 0 5060 sipx.hmherbs.com.
****
****
;; AUTHORITY SECTION:****
hmherbs.com. 1800 IN NS sipx.hmherbs.com.****
****
;; ADDITIONAL SECTION:****
sipx.hmherbs.com. 1800 IN A 70.88.18.153****
****
;; Query time: 48 msec****
;; SERVER: 70.88.18.153#53(70.88.18.153)****
;; WHEN: Sat Jun 16 14:52:03 2012****
;; MSG SIZE rcvd: 105****
****
I've attached the siptrace of the call to 101 I attached in the earlier
logs.****
****
Thanks again,****
Matt****
****
*Sent:* Saturday, June 16, 2012 11:24 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
You are not really providing enough information...****
****
Is the phone local or remote? If it is remote, did you change any of the
phone parameters manually (I ask because the SRV records are missing for
the domain name and that will always cause issues)...****
****
It would really help if you explained the call flow and provided a
siptrace. snippets of logs tell not much...****
****
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
****
wrote:****
Whoops, left the hostname there! Oh well… ****
****
http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel. Per the request of the developer on that page I've attached a INFO level
log of it.****
****
****
*Sent:* Saturday, June 16, 2012 9:39 AM
*Subject:* [sipx-users] voicemail and autoattendant 408 timeout****
****
Hello,****
****
http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331
****
****
I can make calls from extension to extension, however if I let it go to
voicemail the system returns a "408 Timeout" to the phone. Calls directly
to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000
will return "408 Timeout" on the display screen. The CDRs also show the 408
and a failed status. In sipXproxy.log, I see (I've sensored the IP with
xxx.xxx.xxx.xxx) :****
****
"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"****
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write
22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"**
**
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore
OsSocket::write() returned -1, errno = 32"****
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"****
****
Thanks,****
Matt****
****
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sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
****
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Telephone: [phone]
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~~~~~~~~~~~~~~~~~~
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Ask about our Internet Fax services!
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****
LAN/Telephony/Security and Control Systems Helpdesk:****
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****
****
--
~~~~~~~~~~~~~~~~~~
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Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
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Ask about our Internet Fax services!
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****
LAN/Telephony/Security and Control Systems Helpdesk:****
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List Archive: http://list.sipfoundry.org/archive/sipx-users/****
** **
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
** **
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[m***@mattkeys.net]

Fixed now. I just ran through sipxecs-setup again, set the domain for sipx.hmherbs.com, then ran the freeswitch.sh --configtest and --config. It's all good now but the voicemail/operator calls still fail.

[***@sipx ~]# sipxproc --state
{"PageServer"=>"Running",
"PresenceServer"=>"Disabled",
"ConfigAgent"=>"Disabled",
"sipXrecording"=>"Running",
"SharedAppearanceAgent"=>"Running",
"sipXivr"=>"Running",
"SIPStatus"=>"Running",
"ACDServer"=>"Disabled",
"sipXmrtg"=>"Running",
"ParkServer"=>"Running",
"SipXopenfire"=>"Running",
"CallResolver"=>"Running",
"sipXacccode"=>"Running",
"sipXimbot"=>"Running",
"SipXrest"=>"Running",
"ResourceListServer"=>"Running",
"SIPRegistrar"=>"Running",
"SIPXProxy"=>"Running",
"sipXprovision"=>"Running",
"SipXrelay"=>"Running",
"CallResolver-Agent"=>"Disabled",
"SipXbridge"=>"Running",
"FreeSWITCH"=>"Running",
"ConfigServer"=>"Running"}


From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Sunday, June 17, 2012 9:30 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout


Your freeswitch connection test failed. Freeswicth s the media engine for voicemail and auto attendant. Put the log level to debug for media server and restart it and then place a failed call and look at the freswitch logs for why!
On Jun 17, 2012 9:07 AM, "***@mattkeys.net<mailto:***@mattkeys.net>" <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Yes it's every call to voicemail (101), operator (0), or if I call an extension and let it go to voicemail. As you've guessed already this is the initial dev/test installation, so if it would help our troubleshooting I'd be happy to set up an extension for you. The SIP trunks haven't been terminated to it yet so it could only be strictly extension to extension dialing. I couldn't remember if I had sent the server profiles so I've just done that, added intranet domain alias *.sipx.hmherbs.com<http://sipx.hmherbs.com>, *.voice.hmherbs.com<http://voice.hmherbs.com>, and restarted the box. Here's the output you asked for :

[***@sipx sipxpbx]# sipxproc --state
{"FreeSWITCH"=>"ConfigurationTestFailed",
"ConfigServer"=>"Running",
"SipXbridge"=>"Running",
"ConfigAgent"=>"Disabled",
"sipXprovision"=>"Running",
"SipXopenfire"=>"Running",
"PresenceServer"=>"Disabled",
"SipXrelay"=>"Running",
"sipXimbot"=>"Running",
"sipXmrtg"=>"Running",
"SIPRegistrar"=>"Running",
"CallResolver-Agent"=>"Disabled",
"ACDServer"=>"Disabled",
"sipXivr"=>"Running",
"PageServer"=>"Running",
"sipXacccode"=>"Running",
"ParkServer"=>"Running",
"ResourceListServer"=>"Running",
"SharedAppearanceAgent"=>"Running",
"SIPStatus"=>"Running",
"sipXrecording"=>"Running",
"CallResolver"=>"Running",
"SipXrest"=>"Running",
"SIPXProxy"=>"Running"}

From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Sunday, June 17, 2012 6:12 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout


What is the output of
sipxproc --state
Does this happen on every call to the voicemail? When you set the system up did you send the server its profile?

If it were me I'd kick the grand streams to the curb. Since you have them you might be better off to remove the other codecs except g711 from them. I still think something is wring with the domain aspect of it.

Ensure you have a domain alias of the "hostname" and restart services as prompted. When you decide to do a sub domain in production, wipe and rebuild the system from bare metal, don't try to reconfigure it.

Also, check top to make sure the system is not using swap.
On Jun 16, 2012 9:22 PM, "***@mattkeys.net<mailto:***@mattkeys.net>" <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
It's both sides with this problem. The polycom 321s behind the server are pulling programming/provisioning from sipxecs (dhcp and tftp). The grandstream I have here at home I've configured manually. Again, I can successfully make calls to/from all extensions, audio is working bidirectionally, and I see registrations for all of them. The hostname (sipx.hmherbs.com<http://sipx.hmherbs.com>) will probably be a full subdomain soon but at the moment I've just got it set on a A record and the clients pointing to sipxecs for DNS. Under System -> Domain, the domain name is set to "sipx.hmherbs.com<http://sipx.hmherbs.com>", with an alias for voice.hmherbs.com<http://voice.hmherbs.com> and hmherbs.com<http://hmherbs.com> set, depending on what the customer wants to use when we're finished. Under System -> Internet Calling, I've got 192.168.1.0/24<http://192.168.1.0/24>, 192.168.2.0/24<http://192.168.2.0/24>, and 10.1.10.0/24<http://10.1.10.0/24> as Intranet subnets and *.hmherbs.com<http://hmherbs.com> as Intranet domains.

I'm thinking it has to do with the " OsSocket::write() returned -1, errno = 32 " and broken pipe messages... but I didn't know for sure what's causing it.

Thanks,
Matt

From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Saturday, June 16, 2012 5:51 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

its a grandstream having the problem or both?

Can you tell me how the phones were programmed? I think I see the hostname being called "instead of" the domain name. Why would that be?

Did you properly populate the intranet subnets?
On Sat, Jun 16, 2012 at 3:07 PM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Hi Tony, thanks for the quick response. There are phones in both locations. I'm testing using a Grandstream GXV3000 remotely from home but there are two Polycom 321s also registered on the premises. SRV records are there, you've just got to look in the right spot. The grandstream here is pointed to the DNS there, the Polycoms on site use the sipXecs server for DHCP/DNS and gateway :

~]$ dig @70.88.18.153<http://70.88.18.153> SRV _sip._tcp.hmherbs.com<http://tcp.hmherbs.com>

; <<>> DiG 9.7.3-P3-RedHat-9.7.3-8.P3.el6_2.3 <<>> @70.88.18.153<http://70.88.18.153> SRV _sip._tcp.hmherbs.com<http://tcp.hmherbs.com>
; (1 server found)
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1

;; QUESTION SECTION:
;_sip._tcp.hmherbs.com<http://tcp.hmherbs.com>. IN SRV

;; ANSWER SECTION:
_sip._tcp.hmherbs.com<http://tcp.hmherbs.com>. 1800 IN SRV 1 0 5060 sipx.hmherbs.com<http://sipx.hmherbs.com>.

;; AUTHORITY SECTION:
hmherbs.com<http://hmherbs.com>. 1800 IN NS sipx.hmherbs.com<http://sipx.hmherbs.com>.

;; ADDITIONAL SECTION:
sipx.hmherbs.com<http://sipx.hmherbs.com>. 1800 IN A 70.88.18.153

;; Query time: 48 msec
;; SERVER: 70.88.18.153#53(70.88.18.153)
;; WHEN: Sat Jun 16 14:52:03 2012
;; MSG SIZE rcvd: 105

I've attached the siptrace of the call to 101 I attached in the earlier logs.

Thanks again,
Matt

From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Saturday, June 16, 2012 11:24 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

You are not really providing enough information...

Is the phone local or remote? If it is remote, did you change any of the phone parameters manually (I ask because the SRV records are missing for the domain name and that will always cause issues)...

It would really help if you explained the call flow and provided a siptrace. snippets of logs tell not much...

http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
On Sat, Jun 16, 2012 at 10:37 AM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Whoops, left the hostname there! Oh well...

I also found a bug report about this here : http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel . Per the request of the developer on that page I've attached a INFO level log of it.


From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of ***@mattkeys.net<mailto:***@mattkeys.net>
Sent: Saturday, June 16, 2012 9:39 AM
To: sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
Subject: [sipx-users] voicemail and autoattendant 408 timeout

Hello,

I'm having the same problem as this user : http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331

I can make calls from extension to extension, however if I let it go to voicemail the system returns a "408 Timeout" to the phone. Calls directly to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000 will return "408 Timeout" on the display screen. The CDRs also show the 408 and a failed status. In sipXproxy.log, I see (I've sensored the IP with xxx.xxx.xxx.xxx) :

"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction"
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write 22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore OsSocket::write() returned -1, errno = 32"
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction"

Thanks,
Matt


_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

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List Archive: http://list.sipfoundry.org/archive/sipx-users/

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

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List Archive: http://list.sipfoundry.org/archive/sipx-users/

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

[Tony Graziano]

If it were me I would start over with a sub domain if that is your ultimate
goal. It sounds as *though* you have inconsistencies in your setup and will
have issues until you rebuild it.

Post by m***@mattkeys.net
Fixed now. I just ran through sipxecs-setup again, set the domain for
sipx.hmherbs.com, then ran the freeswitch.sh --configtest and --config.
It's all good now but the voicemail/operator calls still fail.****
** **
{"PageServer"=>"Running",****
"PresenceServer"=>"Disabled",****
"ConfigAgent"=>"Disabled",****
"sipXrecording"=>"Running",****
"SharedAppearanceAgent"=>"Running",****
"sipXivr"=>"Running",****
"SIPStatus"=>"Running",****
"ACDServer"=>"Disabled",****
"sipXmrtg"=>"Running",****
"ParkServer"=>"Running",****
"SipXopenfire"=>"Running",****
"CallResolver"=>"Running",****
"sipXacccode"=>"Running",****
"sipXimbot"=>"Running",****
"SipXrest"=>"Running",****
"ResourceListServer"=>"Running",****
"SIPRegistrar"=>"Running",****
"SIPXProxy"=>"Running",****
"sipXprovision"=>"Running",****
"SipXrelay"=>"Running",****
"CallResolver-Agent"=>"Disabled",****
"SipXbridge"=>"Running",****
"FreeSWITCH"=>"Running",****
"ConfigServer"=>"Running"}****
** **
** **
*Sent:* Sunday, June 17, 2012 9:30 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
** **
Your freeswitch connection test failed. Freeswicth s the media engine for
voicemail and auto attendant. Put the log level to debug for media server
and restart it and then place a failed call and look at the freswitch logs
for why!****
*
Yes it's every call to voicemail (101), operator (0), or if I call an
extension and let it go to voicemail. As you've guessed already this is the
initial dev/test installation, so if it would help our troubleshooting I'd
be happy to set up an extension for you. The SIP trunks haven't been
terminated to it yet so it could only be strictly extension to extension
dialing. I couldn't remember if I had sent the server profiles so I've just
done that, added intranet domain alias *.sipx.hmherbs.com, *.
voice.hmherbs.com, and restarted the box. Here's the output you asked for
:****
****
{"FreeSWITCH"=>"ConfigurationTestFailed",****
"ConfigServer"=>"Running",****
"SipXbridge"=>"Running",****
"ConfigAgent"=>"Disabled",****
"sipXprovision"=>"Running",****
"SipXopenfire"=>"Running",****
"PresenceServer"=>"Disabled",****
"SipXrelay"=>"Running",****
"sipXimbot"=>"Running",****
"sipXmrtg"=>"Running",****
"SIPRegistrar"=>"Running",****
"CallResolver-Agent"=>"Disabled",****
"ACDServer"=>"Disabled",****
"sipXivr"=>"Running",****
"PageServer"=>"Running",****
"sipXacccode"=>"Running",****
"ParkServer"=>"Running",****
"ResourceListServer"=>"Running",****
"SharedAppearanceAgent"=>"Running",****
"SIPStatus"=>"Running",****
"sipXrecording"=>"Running",****
"CallResolver"=>"Running",****
"SipXrest"=>"Running",****
"SIPXProxy"=>"Running"}****
****
*Sent:* Sunday, June 17, 2012 6:12 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
What is the output of
sipxproc --state
Does this happen on every call to the voicemail? When you set the system
up did you send the server its profile?****
If it were me I'd kick the grand streams to the curb. Since you have them
you might be better off to remove the other codecs except g711 from them. I
still think something is wring with the domain aspect of it.****
Ensure you have a domain alias of the "hostname" and restart services as
prompted. When you decide to do a sub domain in production, wipe and
rebuild the system from bare metal, don't try to reconfigure it.****
Also, check top to make sure the system is not using swap.****
*
It's both sides with this problem. The polycom 321s behind the server are
pulling programming/provisioning from sipxecs (dhcp and tftp). The
grandstream I have here at home I've configured manually. Again, I can
successfully make calls to/from all extensions, audio is working
bidirectionally, and I see registrations for all of them. The hostname (
sipx.hmherbs.com) will probably be a full subdomain soon but at the
moment I've just got it set on a A record and the clients pointing to
sipxecs for DNS. Under System -> Domain, the domain name is set to "
sipx.hmherbs.com", with an alias for voice.hmherbs.com and hmherbs.comset, depending on what the customer wants to use when we're finished. Under
System -> Internet Calling, I've got 192.168.1.0/24, 192.168.2.0/24, and
10.1.10.0/24 as Intranet subnets and *.hmherbs.com as Intranet domains.***
*
****
I'm thinking it has to do with the " OsSocket::write() returned -1, errno
= 32 " and broken pipe messages… but I didn't know for sure what's causing
it.****
****
Thanks,****
Matt****
****
*Sent:* Saturday, June 16, 2012 5:51 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
its a grandstream having the problem or both?****
****
Can you tell me how the phones were programmed? I think I see the hostname
being called "instead of" the domain name. Why would that be? ****
****
Did you properly populate the intranet subnets?****
wrote:****
Hi Tony, thanks for the quick response. There are phones in both
locations. I'm testing using a Grandstream GXV3000 remotely from home but
there are two Polycom 321s also registered on the premises. SRV records are
there, you've just got to look in the right spot. The grandstream here is
pointed to the DNS there, the Polycoms on site use the sipXecs server for
DHCP/DNS and gateway :****
****
****
_sip._tcp.hmherbs.com****
; (1 server found)****
;; global options: +cmd****
;; Got answer:****
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630****
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1***
*
****
;; QUESTION SECTION:****
;_sip._tcp.hmherbs.com. IN SRV****
****
;; ANSWER SECTION:****
_sip._tcp.hmherbs.com. 1800 IN SRV 1 0 5060 sipx.hmherbs.com.
****
****
;; AUTHORITY SECTION:****
hmherbs.com. 1800 IN NS sipx.hmherbs.com.****
****
;; ADDITIONAL SECTION:****
sipx.hmherbs.com. 1800 IN A 70.88.18.153****
****
;; Query time: 48 msec****
;; SERVER: 70.88.18.153#53(70.88.18.153)****
;; WHEN: Sat Jun 16 14:52:03 2012****
;; MSG SIZE rcvd: 105****
****
I've attached the siptrace of the call to 101 I attached in the earlier
logs.****
****
Thanks again,****
Matt****
****
*Sent:* Saturday, June 16, 2012 11:24 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
You are not really providing enough information...****
****
Is the phone local or remote? If it is remote, did you change any of the
phone parameters manually (I ask because the SRV records are missing for
the domain name and that will always cause issues)...****
****
It would really help if you explained the call flow and provided a
siptrace. snippets of logs tell not much...****
****
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
****
wrote:****
Whoops, left the hostname there! Oh well… ****
****
http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel. Per the request of the developer on that page I've attached a INFO level
log of it.****
****
****
*Sent:* Saturday, June 16, 2012 9:39 AM
*Subject:* [sipx-users] voicemail and autoattendant 408 timeout****
****
Hello,****
****
http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331
****
****
I can make calls from extension to extension, however if I let it go to
voicemail the system returns a "408 Timeout" to the phone. Calls directly
to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000
will return "408 Timeout" on the display screen. The CDRs also show the 408
and a failed status. In sipXproxy.log, I see (I've sensored the IP with
xxx.xxx.xxx.xxx) :****
****
"2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"****
"2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write
22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'"**
**
"2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore
OsSocket::write() returned -1, errno = 32"****
"2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"****
****
Thanks,****
Matt****
****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
****
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
****
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
****
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
****
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
****
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
****
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
****
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
** **
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
** **
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

[m***@mattkeys.net]

Tony / All,

I've started over completely from scratch as you suggested, but I was still having this issue. I eventually found the problem, so I'll share in case anybody runs into this. I ran (as root)

$ service sipxecs stop
$ sipxconfig.sh --database drop
$ sipxconfig.sh --database create
$ sipxecs-setup-system

Ran through the installer again. Set hostname as sipx.hmherbs.com, domain as sipx.hmherbs.com. sipxecs is providing DNS and DHCP.

$ reboot

Logged into sipXsupervisor, added the phones and users back in manually. Ran a test call to 101, same thing was happening evident in /var/log/sipxecs/sipXproxy.log :

"2012-06-23T12:13:39.324672Z":175:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"OsSocket::write 26 (70.88.18.153:15060 :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-23T12:13:39.324728Z":176:SIP:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-173]::writeMore OsSocket::write() returned -1, errno = 32"

I knew it had to do something with binding that port. I do have a non-standard interface setup :

[***@sipx ~]# ifconfig
eth0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54
inet addr:70.88.18.153 Bcast:70.88.18.155 Mask:255.255.255.252
inet6 addr: fe80::225:90ff:fe6a:ed54/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:9245 errors:0 dropped:0 overruns:0 frame:0
TX packets:10261 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:1093753 (1.0 MiB) TX bytes:5985658 (5.7 MiB)
Interrupt:177 Memory:fe9e0000-fea00000

eth0:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54
inet addr:10.1.10.11 Bcast:10.1.10.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Interrupt:177 Memory:fe9e0000-fea00000

eth1 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55
inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0
inet6 addr: fe80::225:90ff:fe6a:ed55/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:382 errors:0 dropped:0 overruns:0 frame:0
TX packets:146 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:48535 (47.3 KiB) TX bytes:28435 (27.7 KiB)
Interrupt:169 Memory:feae0000-feb00000

eth1:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55
inet addr:192.168.2.1 Bcast:192.168.2.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Interrupt:169 Memory:feae0000-feb00000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:46014 errors:0 dropped:0 overruns:0 frame:0
TX packets:46014 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:25541472 (24.3 MiB) TX bytes:25541472 (24.3 MiB)

iptables is running a masquerade for the 192.168.2.0/24 subnet (iptables -t nat -A POSTROUTING -o eth0 -s 192.168.2.0/24 -j MASQUERADE), but not restricting anything at all on the sipxecs interfaces. Phones are on the 1.0/24 subnet (eth1), and public WAN (eth0). Netstat ($ netstat -tulpn |grep 15060 ) was showing port 15060 (freeswitch) bound to 10.1.10.11. I disabled IPV6, rebooted... still the same problem. I then manually edited /etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml, and changed this section :

<!--
Defaults changed, mlk 6.23.12
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
-->

<param name="rtp-ip" value="0.0.0.0"/>
<param name="sip-ip" value="0.0.0.0"/>

<!--
Defaults changed, mlk 6.23.12
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>
-->

<param name="ext-rtp-ip" value="70.88.18.153"/>
<param name="ext-sip-ip" value="70.88.18.153"/>

$ service sipxecs restart

I check netstat, now it's listening on 70.88.18.153 port 15060. I dial 101 to test and I don't see the bind error message in sipXproxy.log but I'm still not hearing the voicemail attendant. The freeswitch logs show the call as being answered. I noticed it was looking for the default intranet subnets (duh.. forgot that step earlier), so I went back into sipxsupervisor and changed the subnets to all 10.1.10.0/24, 192.168.1.0/24, and 192.168.2.0/24, restarted, and tested again. Sure enough, it works!

Short and sweet... if you have multiple interfaces, check that freeswitch is binding to the appropriate interface. If it's not, force it to and tell sipxsupervisor about your subnets.

-Matt

[Tony Graziano]

Therein lies your problem. At no time has multiple NIC's ever been
supported or recommended.

I would suggest you drop back down to one NIC. The issue you are having
WILL reintroduce iteself as the system runs, I guarantee it.

Tony / All, ****
** **
I've started over completely from scratch as you suggested, but I was
still having this issue. I eventually found the problem, so I'll share in
case anybody runs into this. I ran (as root)****
** **
$ service sipxecs stop****
$ sipxconfig.sh --database drop****
$ sipxconfig.sh --database create****
$ sipxecs-setup-system****
** **
Ran through the installer again. Set hostname as sipx.hmherbs.com, domain
as sipx.hmherbs.com. sipxecs is providing DNS and DHCP. ****
** **
$ reboot****
** **
Logged into sipXsupervisor, added the phones and users back in manually.
Ran a test call to 101, same thing was happening evident in
/var/log/sipxecs/sipXproxy.log :****
** **
"2012-06-23T12:13:39.324672Z":175:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"OsSocket::write
26 (70.88.18.153:15060 :-1) send returned -1, errno=32 'Broken pipe'"****
"2012-06-23T12:13:39.324728Z":176:SIP:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-173]::writeMore
OsSocket::write() returned -1, errno = 32"****
** **
I knew it had to do something with binding that port. I do have a
non-standard interface setup :****
** **
eth0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54****
inet addr:70.88.18.153 Bcast:70.88.18.155 Mask:255.255.255.252
****
inet6 addr: fe80::225:90ff:fe6a:ed54/64 Scope:Link****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
RX packets:9245 errors:0 dropped:0 overruns:0 frame:0****
TX packets:10261 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:1000****
RX bytes:1093753 (1.0 MiB) TX bytes:5985658 (5.7 MiB)****
Interrupt:177 Memory:fe9e0000-fea00000****
** **
eth0:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54****
inet addr:10.1.10.11 Bcast:10.1.10.255 Mask:255.255.255.0****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
Interrupt:177 Memory:fe9e0000-fea00000****
** **
eth1 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55****
inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0**
**
inet6 addr: fe80::225:90ff:fe6a:ed55/64 Scope:Link****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
RX packets:382 errors:0 dropped:0 overruns:0 frame:0****
TX packets:146 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:1000****
RX bytes:48535 (47.3 KiB) TX bytes:28435 (27.7 KiB)****
Interrupt:169 Memory:feae0000-feb00000****
** **
eth1:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55****
inet addr:192.168.2.1 Bcast:192.168.2.255 Mask:255.255.255.0**
**
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
Interrupt:169 Memory:feae0000-feb00000****
** **
lo Link encap:Local Loopback****
inet addr:127.0.0.1 Mask:255.0.0.0****
inet6 addr: ::1/128 Scope:Host****
UP LOOPBACK RUNNING MTU:16436 Metric:1****
RX packets:46014 errors:0 dropped:0 overruns:0 frame:0****
TX packets:46014 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:0****
RX bytes:25541472 (24.3 MiB) TX bytes:25541472 (24.3 MiB)****
** **
iptables is running a masquerade for the 192.168.2.0/24 subnet (iptables
-t nat -A POSTROUTING -o eth0 -s 192.168.2.0/24 -j MASQUERADE), but not
restricting anything at all on the sipxecs interfaces. Phones are on the
1.0/24 subnet (eth1), and public WAN (eth0). Netstat ($ netstat -tulpn
|grep 15060 ) was showing port 15060 (freeswitch) bound to 10.1.10.11. I
disabled IPV6, rebooted… still the same problem. I then manually edited
/etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml, and changed
this section :****
** **
<!--****
Defaults changed, mlk 6.23.12****
<param name="rtp-ip" value="$${local_ip_v4}"/>****
<param name="sip-ip" value="$${local_ip_v4}"/>****
-->****
** **
<param name="rtp-ip" value="0.0.0.0"/>****
<param name="sip-ip" value="0.0.0.0"/>****
** **
<!--****
Defaults changed, mlk 6.23.12****
<param name="ext-rtp-ip" value="auto-nat"/>****
<param name="ext-sip-ip" value="auto-nat"/>****
-->****
** **
<param name="ext-rtp-ip" value="70.88.18.153"/>****
<param name="ext-sip-ip" value="70.88.18.153"/>****
** **
$ service sipxecs restart****
** **
I check netstat, now it's listening on 70.88.18.153 port 15060. I dial 101
to test and I don't see the bind error message in sipXproxy.log but I'm
still not hearing the voicemail attendant. The freeswitch logs show the
call as being answered. I noticed it was looking for the default intranet
subnets (duh.. forgot that step earlier), so I went back into
sipxsupervisor and changed the subnets to all 10.1.10.0/24, 192.168.1.0/24,
and 192.168.2.0/24, restarted, and tested again. Sure enough, it works!***
*
** **
Short and sweet… if you have multiple interfaces, check that freeswitch is
binding to the appropriate interface. If it's not, force it to and tell
sipxsupervisor about your subnets.****
** **
-Matt****
** **
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>

[m***@mattkeys.net]

So I've noticed! On a reboot it blew away the freeswitch modifications I had entered. I'm curious what would happen if I make them immutable?


From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Saturday, June 23, 2012 12:59 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

Therein lies your problem. At no time has multiple NIC's ever been supported or recommended.

I would suggest you drop back down to one NIC. The issue you are having WILL reintroduce iteself as the system runs, I guarantee it.
On Sat, Jun 23, 2012 at 10:20 AM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Tony / All,

I've started over completely from scratch as you suggested, but I was still having this issue. I eventually found the problem, so I'll share in case anybody runs into this. I ran (as root)

$ service sipxecs stop
$ sipxconfig.sh --database drop
$ sipxconfig.sh --database create
$ sipxecs-setup-system

Ran through the installer again. Set hostname as sipx.hmherbs.com<http://sipx.hmherbs.com>, domain as sipx.hmherbs.com<http://sipx.hmherbs.com>. sipxecs is providing DNS and DHCP.

$ reboot

Logged into sipXsupervisor, added the phones and users back in manually. Ran a test call to 101, same thing was happening evident in /var/log/sipxecs/sipXproxy.log :

"2012-06-23T12:13:39.324672Z":175:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"OsSocket::write 26 (70.88.18.153:15060<http://70.88.18.153:15060> :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-23T12:13:39.324728Z":176:SIP:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-173]::writeMore OsSocket::write() returned -1, errno = 32"

I knew it had to do something with binding that port. I do have a non-standard interface setup :

[***@sipx ~]# ifconfig
eth0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54
inet addr:70.88.18.153 Bcast:70.88.18.155 Mask:255.255.255.252
inet6 addr: fe80::225:90ff:fe6a:ed54/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:9245 errors:0 dropped:0 overruns:0 frame:0
TX packets:10261 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:1093753 (1.0 MiB) TX bytes:5985658 (5.7 MiB)
Interrupt:177 Memory:fe9e0000-fea00000

eth0:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54
inet addr:10.1.10.11 Bcast:10.1.10.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Interrupt:177 Memory:fe9e0000-fea00000

eth1 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55
inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0
inet6 addr: fe80::225:90ff:fe6a:ed55/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:382 errors:0 dropped:0 overruns:0 frame:0
TX packets:146 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:48535 (47.3 KiB) TX bytes:28435 (27.7 KiB)
Interrupt:169 Memory:feae0000-feb00000

eth1:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55
inet addr:192.168.2.1 Bcast:192.168.2.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Interrupt:169 Memory:feae0000-feb00000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:46014 errors:0 dropped:0 overruns:0 frame:0
TX packets:46014 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:25541472 (24.3 MiB) TX bytes:25541472 (24.3 MiB)

iptables is running a masquerade for the 192.168.2.0/24<http://192.168.2.0/24> subnet (iptables -t nat -A POSTROUTING -o eth0 -s 192.168.2.0/24<http://192.168.2.0/24> -j MASQUERADE), but not restricting anything at all on the sipxecs interfaces. Phones are on the 1.0/24 subnet (eth1), and public WAN (eth0). Netstat ($ netstat -tulpn |grep 15060 ) was showing port 15060 (freeswitch) bound to 10.1.10.11. I disabled IPV6, rebooted... still the same problem. I then manually edited /etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml, and changed this section :

<!--
Defaults changed, mlk 6.23.12
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
-->

<param name="rtp-ip" value="0.0.0.0"/>
<param name="sip-ip" value="0.0.0.0"/>

<!--
Defaults changed, mlk 6.23.12
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>
-->

<param name="ext-rtp-ip" value="70.88.18.153"/>
<param name="ext-sip-ip" value="70.88.18.153"/>

$ service sipxecs restart

I check netstat, now it's listening on 70.88.18.153 port 15060. I dial 101 to test and I don't see the bind error message in sipXproxy.log but I'm still not hearing the voicemail attendant. The freeswitch logs show the call as being answered. I noticed it was looking for the default intranet subnets (duh.. forgot that step earlier), so I went back into sipxsupervisor and changed the subnets to all 10.1.10.0/24<http://10.1.10.0/24>, 192.168.1.0/24<http://192.168.1.0/24>, and 192.168.2.0/24<http://192.168.2.0/24>, restarted, and tested again. Sure enough, it works!

Short and sweet... if you have multiple interfaces, check that freeswitch is binding to the appropriate interface. If it's not, force it to and tell sipxsupervisor about your subnets.

-Matt


_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!<http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>


LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

[Tony Graziano]

You really need to understand how sipx works with projecting and managing
configuration files. There is a way to do this via the .vm files but you
will need to take care to understand how to deal with the upgrade/update
process as well.

Post by m***@mattkeys.net
So I've noticed! On a reboot it blew away the freeswitch modifications I
had entered. I'm curious what would happen if I make them immutable?****
** **
** **
*Sent:* Saturday, June 23, 2012 12:59 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
** **
Therein lies your problem. At no time has multiple NIC's ever been
supported or recommended.****
** **
I would suggest you drop back down to one NIC. The issue you are having
WILL reintroduce iteself as the system runs, I guarantee it.****
wrote:****
Tony / All, ****
****
I've started over completely from scratch as you suggested, but I was
still having this issue. I eventually found the problem, so I'll share in
case anybody runs into this. I ran (as root)****
****
$ service sipxecs stop****
$ sipxconfig.sh --database drop****
$ sipxconfig.sh --database create****
$ sipxecs-setup-system****
****
Ran through the installer again. Set hostname as sipx.hmherbs.com, domain
as sipx.hmherbs.com. sipxecs is providing DNS and DHCP. ****
****
$ reboot****
****
Logged into sipXsupervisor, added the phones and users back in manually.
Ran a test call to 101, same thing was happening evident in
/var/log/sipxecs/sipXproxy.log :****
****
"2012-06-23T12:13:39.324672Z":175:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"OsSocket::write
26 (70.88.18.153:15060 :-1) send returned -1, errno=32 'Broken pipe'"****
"2012-06-23T12:13:39.324728Z":176:SIP:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-173]::writeMore
OsSocket::write() returned -1, errno = 32"****
****
I knew it had to do something with binding that port. I do have a
non-standard interface setup :****
****
eth0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54****
inet addr:70.88.18.153 Bcast:70.88.18.155 Mask:255.255.255.252
****
inet6 addr: fe80::225:90ff:fe6a:ed54/64 Scope:Link****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
RX packets:9245 errors:0 dropped:0 overruns:0 frame:0****
TX packets:10261 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:1000****
RX bytes:1093753 (1.0 MiB) TX bytes:5985658 (5.7 MiB)****
Interrupt:177 Memory:fe9e0000-fea00000****
****
eth0:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54****
inet addr:10.1.10.11 Bcast:10.1.10.255 Mask:255.255.255.0****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
Interrupt:177 Memory:fe9e0000-fea00000****
****
eth1 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55****
inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0**
**
inet6 addr: fe80::225:90ff:fe6a:ed55/64 Scope:Link****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
RX packets:382 errors:0 dropped:0 overruns:0 frame:0****
TX packets:146 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:1000****
RX bytes:48535 (47.3 KiB) TX bytes:28435 (27.7 KiB)****
Interrupt:169 Memory:feae0000-feb00000****
****
eth1:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55****
inet addr:192.168.2.1 Bcast:192.168.2.255 Mask:255.255.255.0**
**
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
Interrupt:169 Memory:feae0000-feb00000****
****
lo Link encap:Local Loopback****
inet addr:127.0.0.1 Mask:255.0.0.0****
inet6 addr: ::1/128 Scope:Host****
UP LOOPBACK RUNNING MTU:16436 Metric:1****
RX packets:46014 errors:0 dropped:0 overruns:0 frame:0****
TX packets:46014 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:0****
RX bytes:25541472 (24.3 MiB) TX bytes:25541472 (24.3 MiB)****
****
iptables is running a masquerade for the 192.168.2.0/24 subnet (iptables
-t nat -A POSTROUTING -o eth0 -s 192.168.2.0/24 -j MASQUERADE), but not
restricting anything at all on the sipxecs interfaces. Phones are on the
1.0/24 subnet (eth1), and public WAN (eth0). Netstat ($ netstat -tulpn
|grep 15060 ) was showing port 15060 (freeswitch) bound to 10.1.10.11. I
disabled IPV6, rebooted… still the same problem. I then manually edited
/etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml, and changed
this section :****
****
<!--****
Defaults changed, mlk 6.23.12****
<param name="rtp-ip" value="$${local_ip_v4}"/>****
<param name="sip-ip" value="$${local_ip_v4}"/>****
-->****
****
<param name="rtp-ip" value="0.0.0.0"/>****
<param name="sip-ip" value="0.0.0.0"/>****
****
<!--****
Defaults changed, mlk 6.23.12****
<param name="ext-rtp-ip" value="auto-nat"/>****
<param name="ext-sip-ip" value="auto-nat"/>****
-->****
****
<param name="ext-rtp-ip" value="70.88.18.153"/>****
<param name="ext-sip-ip" value="70.88.18.153"/>****
****
$ service sipxecs restart****
****
I check netstat, now it's listening on 70.88.18.153 port 15060. I dial 101
to test and I don't see the bind error message in sipXproxy.log but I'm
still not hearing the voicemail attendant. The freeswitch logs show the
call as being answered. I noticed it was looking for the default intranet
subnets (duh.. forgot that step earlier), so I went back into
sipxsupervisor and changed the subnets to all 10.1.10.0/24, 192.168.1.0/24,
and 192.168.2.0/24, restarted, and tested again. Sure enough, it works!***
*
****
Short and sweet… if you have multiple interfaces, check that freeswitch is
binding to the appropriate interface. If it's not, force it to and tell
sipxsupervisor about your subnets.****
****
-Matt****
****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
** **
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
** **
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>****
** **
** **
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
** **
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>

[m***@mattkeys.net]

I'd love to know more about this, as I see that this issue has been "fixed" ( http://track.sipfoundry.org/browse/XX-8692 ) but I haven't run across any documentation to dive deep into it. I've played with 4.6 beta (when is the expected stable release btw?) a month or so ago before deploying 4.4. I remember seeing in sipxsupervisor that it had iptables/firewall support. Will the 4.6 version be able to bond to specified interfaces?

Thanks,
Matt


From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Saturday, June 23, 2012 4:35 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

You really need to understand how sipx works with projecting and managing configuration files. There is a way to do this via the .vm files but you will need to take care to understand how to deal with the upgrade/update process as well.
On Sat, Jun 23, 2012 at 1:27 PM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
So I've noticed! On a reboot it blew away the freeswitch modifications I had entered. I'm curious what would happen if I make them immutable?


From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Saturday, June 23, 2012 12:59 PM

To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

Therein lies your problem. At no time has multiple NIC's ever been supported or recommended.

I would suggest you drop back down to one NIC. The issue you are having WILL reintroduce iteself as the system runs, I guarantee it.
On Sat, Jun 23, 2012 at 10:20 AM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Tony / All,

I've started over completely from scratch as you suggested, but I was still having this issue. I eventually found the problem, so I'll share in case anybody runs into this. I ran (as root)

$ service sipxecs stop
$ sipxconfig.sh --database drop
$ sipxconfig.sh --database create
$ sipxecs-setup-system

Ran through the installer again. Set hostname as sipx.hmherbs.com<http://sipx.hmherbs.com>, domain as sipx.hmherbs.com<http://sipx.hmherbs.com>. sipxecs is providing DNS and DHCP.

$ reboot

Logged into sipXsupervisor, added the phones and users back in manually. Ran a test call to 101, same thing was happening evident in /var/log/sipxecs/sipXproxy.log :

"2012-06-23T12:13:39.324672Z":175:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"OsSocket::write 26 (70.88.18.153:15060<http://70.88.18.153:15060> :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-23T12:13:39.324728Z":176:SIP:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-173]::writeMore OsSocket::write() returned -1, errno = 32"

I knew it had to do something with binding that port. I do have a non-standard interface setup :

[***@sipx ~]# ifconfig
eth0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54
inet addr:70.88.18.153 Bcast:70.88.18.155 Mask:255.255.255.252
inet6 addr: fe80::225:90ff:fe6a:ed54/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:9245 errors:0 dropped:0 overruns:0 frame:0
TX packets:10261 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:1093753 (1.0 MiB) TX bytes:5985658 (5.7 MiB)
Interrupt:177 Memory:fe9e0000-fea00000

eth0:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54
inet addr:10.1.10.11 Bcast:10.1.10.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Interrupt:177 Memory:fe9e0000-fea00000

eth1 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55
inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0
inet6 addr: fe80::225:90ff:fe6a:ed55/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:382 errors:0 dropped:0 overruns:0 frame:0
TX packets:146 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:48535 (47.3 KiB) TX bytes:28435 (27.7 KiB)
Interrupt:169 Memory:feae0000-feb00000

eth1:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55
inet addr:192.168.2.1 Bcast:192.168.2.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Interrupt:169 Memory:feae0000-feb00000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:46014 errors:0 dropped:0 overruns:0 frame:0
TX packets:46014 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:25541472 (24.3 MiB) TX bytes:25541472 (24.3 MiB)

iptables is running a masquerade for the 192.168.2.0/24<http://192.168.2.0/24> subnet (iptables -t nat -A POSTROUTING -o eth0 -s 192.168.2.0/24<http://192.168.2.0/24> -j MASQUERADE), but not restricting anything at all on the sipxecs interfaces. Phones are on the 1.0/24 subnet (eth1), and public WAN (eth0). Netstat ($ netstat -tulpn |grep 15060 ) was showing port 15060 (freeswitch) bound to 10.1.10.11. I disabled IPV6, rebooted... still the same problem. I then manually edited /etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml, and changed this section :

<!--
Defaults changed, mlk 6.23.12
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
-->

<param name="rtp-ip" value="0.0.0.0"/>
<param name="sip-ip" value="0.0.0.0"/>

<!--
Defaults changed, mlk 6.23.12
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>
-->

<param name="ext-rtp-ip" value="70.88.18.153"/>
<param name="ext-sip-ip" value="70.88.18.153"/>

$ service sipxecs restart

I check netstat, now it's listening on 70.88.18.153 port 15060. I dial 101 to test and I don't see the bind error message in sipXproxy.log but I'm still not hearing the voicemail attendant. The freeswitch logs show the call as being answered. I noticed it was looking for the default intranet subnets (duh.. forgot that step earlier), so I went back into sipxsupervisor and changed the subnets to all 10.1.10.0/24<http://10.1.10.0/24>, 192.168.1.0/24<http://192.168.1.0/24>, and 192.168.2.0/24<http://192.168.2.0/24>, restarted, and tested again. Sure enough, it works!

Short and sweet... if you have multiple interfaces, check that freeswitch is binding to the appropriate interface. If it's not, force it to and tell sipxsupervisor about your subnets.

-Matt


_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!<http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>


LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!<http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

[Michael Picher]

You can bond interfaces today... As long as the interfaces present
themselves to the Linux system as a single network interface. You can then
have connections to multiple switches for a diverse path.

Thanks,
Mike

Post by m***@mattkeys.net
I'd love to know more about this, as I see that this issue has been
"fixed" ( http://track.sipfoundry.org/browse/XX-8692 ) but I haven't run
across any documentation to dive deep into it. I've played with 4.6 beta
(when is the expected stable release btw?) a month or so ago before
deploying 4.4. I remember seeing in sipxsupervisor that it had
iptables/firewall support. Will the 4.6 version be able to bond to
specified interfaces?****
** **
Thanks,****
Matt****
** **
** **
*Sent:* Saturday, June 23, 2012 4:35 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
** **
You really need to understand how sipx works with projecting and managing
configuration files. There is a way to do this via the .vm files but you
will need to take care to understand how to deal with the upgrade/update
process as well.****
wrote:****
So I've noticed! On a reboot it blew away the freeswitch modifications I
had entered. I'm curious what would happen if I make them immutable?****
****
****
*Sent:* Saturday, June 23, 2012 12:59 PM****
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
Therein lies your problem. At no time has multiple NIC's ever been
supported or recommended.****
****
I would suggest you drop back down to one NIC. The issue you are having
WILL reintroduce iteself as the system runs, I guarantee it.****
wrote:****
Tony / All, ****
****
I've started over completely from scratch as you suggested, but I was
still having this issue. I eventually found the problem, so I'll share in
case anybody runs into this. I ran (as root)****
****
$ service sipxecs stop****
$ sipxconfig.sh --database drop****
$ sipxconfig.sh --database create****
$ sipxecs-setup-system****
****
Ran through the installer again. Set hostname as sipx.hmherbs.com, domain
as sipx.hmherbs.com. sipxecs is providing DNS and DHCP. ****
****
$ reboot****
****
Logged into sipXsupervisor, added the phones and users back in manually.
Ran a test call to 101, same thing was happening evident in
/var/log/sipxecs/sipXproxy.log :****
****
"2012-06-23T12:13:39.324672Z":175:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"OsSocket::write
26 (70.88.18.153:15060 :-1) send returned -1, errno=32 'Broken pipe'"****
"2012-06-23T12:13:39.324728Z":176:SIP:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-173]::writeMore
OsSocket::write() returned -1, errno = 32"****
****
I knew it had to do something with binding that port. I do have a
non-standard interface setup :****
****
eth0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54****
inet addr:70.88.18.153 Bcast:70.88.18.155 Mask:255.255.255.252
****
inet6 addr: fe80::225:90ff:fe6a:ed54/64 Scope:Link****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
RX packets:9245 errors:0 dropped:0 overruns:0 frame:0****
TX packets:10261 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:1000****
RX bytes:1093753 (1.0 MiB) TX bytes:5985658 (5.7 MiB)****
Interrupt:177 Memory:fe9e0000-fea00000****
****
eth0:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54****
inet addr:10.1.10.11 Bcast:10.1.10.255 Mask:255.255.255.0****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
Interrupt:177 Memory:fe9e0000-fea00000****
****
eth1 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55****
inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0**
**
inet6 addr: fe80::225:90ff:fe6a:ed55/64 Scope:Link****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
RX packets:382 errors:0 dropped:0 overruns:0 frame:0****
TX packets:146 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:1000****
RX bytes:48535 (47.3 KiB) TX bytes:28435 (27.7 KiB)****
Interrupt:169 Memory:feae0000-feb00000****
****
eth1:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55****
inet addr:192.168.2.1 Bcast:192.168.2.255 Mask:255.255.255.0**
**
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
Interrupt:169 Memory:feae0000-feb00000****
****
lo Link encap:Local Loopback****
inet addr:127.0.0.1 Mask:255.0.0.0****
inet6 addr: ::1/128 Scope:Host****
UP LOOPBACK RUNNING MTU:16436 Metric:1****
RX packets:46014 errors:0 dropped:0 overruns:0 frame:0****
TX packets:46014 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:0****
RX bytes:25541472 (24.3 MiB) TX bytes:25541472 (24.3 MiB)****
****
iptables is running a masquerade for the 192.168.2.0/24 subnet (iptables
-t nat -A POSTROUTING -o eth0 -s 192.168.2.0/24 -j MASQUERADE), but not
restricting anything at all on the sipxecs interfaces. Phones are on the
1.0/24 subnet (eth1), and public WAN (eth0). Netstat ($ netstat -tulpn
|grep 15060 ) was showing port 15060 (freeswitch) bound to 10.1.10.11. I
disabled IPV6, rebooted… still the same problem. I then manually edited
/etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml, and changed
this section :****
****
<!--****
Defaults changed, mlk 6.23.12****
<param name="rtp-ip" value="$${local_ip_v4}"/>****
<param name="sip-ip" value="$${local_ip_v4}"/>****
-->****
****
<param name="rtp-ip" value="0.0.0.0"/>****
<param name="sip-ip" value="0.0.0.0"/>****
****
<!--****
Defaults changed, mlk 6.23.12****
<param name="ext-rtp-ip" value="auto-nat"/>****
<param name="ext-sip-ip" value="auto-nat"/>****
-->****
****
<param name="ext-rtp-ip" value="70.88.18.153"/>****
<param name="ext-sip-ip" value="70.88.18.153"/>****
****
$ service sipxecs restart****
****
I check netstat, now it's listening on 70.88.18.153 port 15060. I dial 101
to test and I don't see the bind error message in sipXproxy.log but I'm
still not hearing the voicemail attendant. The freeswitch logs show the
call as being answered. I noticed it was looking for the default intranet
subnets (duh.. forgot that step earlier), so I went back into
sipxsupervisor and changed the subnets to all 10.1.10.0/24, 192.168.1.0/24,
and 192.168.2.0/24, restarted, and tested again. Sure enough, it works!***
*
****
Short and sweet… if you have multiple interfaces, check that freeswitch is
binding to the appropriate interface. If it's not, force it to and tell
sipxsupervisor about your subnets.****
****
-Matt****
****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
****
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
****
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>****
****
****
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
****
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
** **
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
** **
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>****
** **
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
** **
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Blog: http://blog.myitdepartment.net****
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--
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eZuce, Inc.

300 Brickstone Square****

Suite 201****

Andover, MA. 01810
O.978-296-1005 X2015
M.207-956-0262
@mpicher <http://twitter.com/mpicher>
linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
www.ezuce.com

------------------------------------------------------------------------------------------------------------
There are 10 kinds of people in the world, those who understand binary and
those who don't.

[m***@mattkeys.net]

So with a LACP (802.3ad) trunk presented as bond0 I can throw as many as I want at it?


From: sipx-users-***@list.sipfoundry.org [mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Sunday, June 24, 2012 5:01 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

You can bond interfaces today... As long as the interfaces present themselves to the Linux system as a single network interface. You can then have connections to multiple switches for a diverse path.

Thanks,
Mike

On Sun, Jun 24, 2012 at 12:00 AM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
I'd love to know more about this, as I see that this issue has been "fixed" ( http://track.sipfoundry.org/browse/XX-8692 ) but I haven't run across any documentation to dive deep into it. I've played with 4.6 beta (when is the expected stable release btw?) a month or so ago before deploying 4.4. I remember seeing in sipxsupervisor that it had iptables/firewall support. Will the 4.6 version be able to bond to specified interfaces?

Thanks,
Matt


From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Saturday, June 23, 2012 4:35 PM

To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

You really need to understand how sipx works with projecting and managing configuration files. There is a way to do this via the .vm files but you will need to take care to understand how to deal with the upgrade/update process as well.
On Sat, Jun 23, 2012 at 1:27 PM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
So I've noticed! On a reboot it blew away the freeswitch modifications I had entered. I'm curious what would happen if I make them immutable?


From: sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org> [mailto:sipx-users-***@list.sipfoundry.org<mailto:sipx-users-***@list.sipfoundry.org>] On Behalf Of Tony Graziano
Sent: Saturday, June 23, 2012 12:59 PM

To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout

Therein lies your problem. At no time has multiple NIC's ever been supported or recommended.

I would suggest you drop back down to one NIC. The issue you are having WILL reintroduce iteself as the system runs, I guarantee it.
On Sat, Jun 23, 2012 at 10:20 AM, ***@mattkeys.net<mailto:***@mattkeys.net> <***@mattkeys.net<mailto:***@mattkeys.net>> wrote:
Tony / All,

I've started over completely from scratch as you suggested, but I was still having this issue. I eventually found the problem, so I'll share in case anybody runs into this. I ran (as root)

$ service sipxecs stop
$ sipxconfig.sh --database drop
$ sipxconfig.sh --database create
$ sipxecs-setup-system

Ran through the installer again. Set hostname as sipx.hmherbs.com<http://sipx.hmherbs.com>, domain as sipx.hmherbs.com<http://sipx.hmherbs.com>. sipxecs is providing DNS and DHCP.

$ reboot

Logged into sipXsupervisor, added the phones and users back in manually. Ran a test call to 101, same thing was happening evident in /var/log/sipxecs/sipXproxy.log :

"2012-06-23T12:13:39.324672Z":175:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"OsSocket::write 26 (70.88.18.153:15060<http://70.88.18.153:15060> :-1) send returned -1, errno=32 'Broken pipe'"
"2012-06-23T12:13:39.324728Z":176:SIP:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-173]::writeMore OsSocket::write() returned -1, errno = 32"

I knew it had to do something with binding that port. I do have a non-standard interface setup :

[***@sipx ~]# ifconfig
eth0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54
inet addr:70.88.18.153 Bcast:70.88.18.155 Mask:255.255.255.252
inet6 addr: fe80::225:90ff:fe6a:ed54/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:9245 errors:0 dropped:0 overruns:0 frame:0
TX packets:10261 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:1093753 (1.0 MiB) TX bytes:5985658 (5.7 MiB)
Interrupt:177 Memory:fe9e0000-fea00000

eth0:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54
inet addr:10.1.10.11 Bcast:10.1.10.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Interrupt:177 Memory:fe9e0000-fea00000

eth1 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55
inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0
inet6 addr: fe80::225:90ff:fe6a:ed55/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:382 errors:0 dropped:0 overruns:0 frame:0
TX packets:146 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:48535 (47.3 KiB) TX bytes:28435 (27.7 KiB)
Interrupt:169 Memory:feae0000-feb00000

eth1:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55
inet addr:192.168.2.1 Bcast:192.168.2.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Interrupt:169 Memory:feae0000-feb00000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:46014 errors:0 dropped:0 overruns:0 frame:0
TX packets:46014 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:25541472 (24.3 MiB) TX bytes:25541472 (24.3 MiB)

iptables is running a masquerade for the 192.168.2.0/24<http://192.168.2.0/24> subnet (iptables -t nat -A POSTROUTING -o eth0 -s 192.168.2.0/24<http://192.168.2.0/24> -j MASQUERADE), but not restricting anything at all on the sipxecs interfaces. Phones are on the 1.0/24 subnet (eth1), and public WAN (eth0). Netstat ($ netstat -tulpn |grep 15060 ) was showing port 15060 (freeswitch) bound to 10.1.10.11. I disabled IPV6, rebooted... still the same problem. I then manually edited /etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml, and changed this section :

<!--
Defaults changed, mlk 6.23.12
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
-->

<param name="rtp-ip" value="0.0.0.0"/>
<param name="sip-ip" value="0.0.0.0"/>

<!--
Defaults changed, mlk 6.23.12
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>
-->

<param name="ext-rtp-ip" value="70.88.18.153"/>
<param name="ext-sip-ip" value="70.88.18.153"/>

$ service sipxecs restart

I check netstat, now it's listening on 70.88.18.153 port 15060. I dial 101 to test and I don't see the bind error message in sipXproxy.log but I'm still not hearing the voicemail attendant. The freeswitch logs show the call as being answered. I noticed it was looking for the default intranet subnets (duh.. forgot that step earlier), so I went back into sipxsupervisor and changed the subnets to all 10.1.10.0/24<http://10.1.10.0/24>, 192.168.1.0/24<http://192.168.1.0/24>, and 192.168.2.0/24<http://192.168.2.0/24>, restarted, and tested again. Sure enough, it works!

Short and sweet... if you have multiple interfaces, check that freeswitch is binding to the appropriate interface. If it's not, force it to and tell sipxsupervisor about your subnets.

-Matt


_______________________________________________
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sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]<tel:[phone]>
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]<tel:[phone]>
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!<http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>


LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]<tel:[phone]>
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

_______________________________________________
sipx-users mailing list
sipx-***@list.sipfoundry.org<mailto:sipx-***@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]<tel:[phone]>
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>
Fax: [phone]<tel:[phone]>
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!<http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: [phone]<tel:[phone]>
sip: ***@voice.myitdepartment.net<mailto:***@voice.myitdepartment.net>

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net

_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/



--
Michael Picher, Director of Technical Services
eZuce, Inc.

300 Brickstone Square

Suite 201

Andover, MA. 01810
O.978-296-1005 X2015
M.207-956-0262
@mpicher <http://twitter.com/mpicher>
linkedin<http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
www.ezuce.com<http://www.ezuce.com>

------------------------------------------------------------------------------------------------------------
There are 10 kinds of people in the world, those who understand binary and those who don't.

[Michael Picher]

Ayuh...

Post by m***@mattkeys.net
So with a LACP (802.3ad) trunk presented as bond0 I can throw as many as I
want at it?****
** **
** **
*Sent:* Sunday, June 24, 2012 5:01 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
** **
You can bond interfaces today... As long as the interfaces present
themselves to the Linux system as a single network interface. You can then
have connections to multiple switches for a diverse path.****
** **
Thanks,****
Mike****
** **
wrote:****
I'd love to know more about this, as I see that this issue has been
"fixed" ( http://track.sipfoundry.org/browse/XX-8692 ) but I haven't run
across any documentation to dive deep into it. I've played with 4.6 beta
(when is the expected stable release btw?) a month or so ago before
deploying 4.4. I remember seeing in sipxsupervisor that it had
iptables/firewall support. Will the 4.6 version be able to bond to
specified interfaces?****
****
Thanks,****
Matt****
****
****
*Sent:* Saturday, June 23, 2012 4:35 PM****
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
You really need to understand how sipx works with projecting and managing
configuration files. There is a way to do this via the .vm files but you
will need to take care to understand how to deal with the upgrade/update
process as well.****
wrote:****
So I've noticed! On a reboot it blew away the freeswitch modifications I
had entered. I'm curious what would happen if I make them immutable?****
****
****
*Sent:* Saturday, June 23, 2012 12:59 PM****
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
****
Therein lies your problem. At no time has multiple NIC's ever been
supported or recommended.****
****
I would suggest you drop back down to one NIC. The issue you are having
WILL reintroduce iteself as the system runs, I guarantee it.****
wrote:****
Tony / All, ****
****
I've started over completely from scratch as you suggested, but I was
still having this issue. I eventually found the problem, so I'll share in
case anybody runs into this. I ran (as root)****
****
$ service sipxecs stop****
$ sipxconfig.sh --database drop****
$ sipxconfig.sh --database create****
$ sipxecs-setup-system****
****
Ran through the installer again. Set hostname as sipx.hmherbs.com, domain
as sipx.hmherbs.com. sipxecs is providing DNS and DHCP. ****
****
$ reboot****
****
Logged into sipXsupervisor, added the phones and users back in manually.
Ran a test call to 101, same thing was happening evident in
/var/log/sipxecs/sipXproxy.log :****
****
"2012-06-23T12:13:39.324672Z":175:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"OsSocket::write
26 (70.88.18.153:15060 :-1) send returned -1, errno=32 'Broken pipe'"****
"2012-06-23T12:13:39.324728Z":176:SIP:ERR:sipx.hmherbs.com:SipClientTcp-173:41585940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-173]::writeMore
OsSocket::write() returned -1, errno = 32"****
****
I knew it had to do something with binding that port. I do have a
non-standard interface setup :****
****
eth0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54****
inet addr:70.88.18.153 Bcast:70.88.18.155 Mask:255.255.255.252
****
inet6 addr: fe80::225:90ff:fe6a:ed54/64 Scope:Link****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
RX packets:9245 errors:0 dropped:0 overruns:0 frame:0****
TX packets:10261 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:1000****
RX bytes:1093753 (1.0 MiB) TX bytes:5985658 (5.7 MiB)****
Interrupt:177 Memory:fe9e0000-fea00000****
****
eth0:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:54****
inet addr:10.1.10.11 Bcast:10.1.10.255 Mask:255.255.255.0****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
Interrupt:177 Memory:fe9e0000-fea00000****
****
eth1 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55****
inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0**
**
inet6 addr: fe80::225:90ff:fe6a:ed55/64 Scope:Link****
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
RX packets:382 errors:0 dropped:0 overruns:0 frame:0****
TX packets:146 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:1000****
RX bytes:48535 (47.3 KiB) TX bytes:28435 (27.7 KiB)****
Interrupt:169 Memory:feae0000-feb00000****
****
eth1:0 Link encap:Ethernet HWaddr 00:25:90:6A:ED:55****
inet addr:192.168.2.1 Bcast:192.168.2.255 Mask:255.255.255.0**
**
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1****
Interrupt:169 Memory:feae0000-feb00000****
****
lo Link encap:Local Loopback****
inet addr:127.0.0.1 Mask:255.0.0.0****
inet6 addr: ::1/128 Scope:Host****
UP LOOPBACK RUNNING MTU:16436 Metric:1****
RX packets:46014 errors:0 dropped:0 overruns:0 frame:0****
TX packets:46014 errors:0 dropped:0 overruns:0 carrier:0****
collisions:0 txqueuelen:0****
RX bytes:25541472 (24.3 MiB) TX bytes:25541472 (24.3 MiB)****
****
iptables is running a masquerade for the 192.168.2.0/24 subnet (iptables
-t nat -A POSTROUTING -o eth0 -s 192.168.2.0/24 -j MASQUERADE), but not
restricting anything at all on the sipxecs interfaces. Phones are on the
1.0/24 subnet (eth1), and public WAN (eth0). Netstat ($ netstat -tulpn
|grep 15060 ) was showing port 15060 (freeswitch) bound to 10.1.10.11. I
disabled IPV6, rebooted… still the same problem. I then manually edited
/etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml, and changed
this section :****
****
<!--****
Defaults changed, mlk 6.23.12****
<param name="rtp-ip" value="$${local_ip_v4}"/>****
<param name="sip-ip" value="$${local_ip_v4}"/>****
-->****
****
<param name="rtp-ip" value="0.0.0.0"/>****
<param name="sip-ip" value="0.0.0.0"/>****
****
<!--****
Defaults changed, mlk 6.23.12****
<param name="ext-rtp-ip" value="auto-nat"/>****
<param name="ext-sip-ip" value="auto-nat"/>****
-->****
****
<param name="ext-rtp-ip" value="70.88.18.153"/>****
<param name="ext-sip-ip" value="70.88.18.153"/>****
****
$ service sipxecs restart****
****
I check netstat, now it's listening on 70.88.18.153 port 15060. I dial 101
to test and I don't see the bind error message in sipXproxy.log but I'm
still not hearing the voicemail attendant. The freeswitch logs show the
call as being answered. I noticed it was looking for the default intranet
subnets (duh.. forgot that step earlier), so I went back into
sipxsupervisor and changed the subnets to all 10.1.10.0/24, 192.168.1.0/24,
and 192.168.2.0/24, restarted, and tested again. Sure enough, it works!***
*
****
Short and sweet… if you have multiple interfaces, check that freeswitch is
binding to the appropriate interface. If it's not, force it to and tell
sipxsupervisor about your subnets.****
****
-Matt****
****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
****
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
****
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>****
****
****
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
****
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
****
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~****
****
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013%22>****
****
LAN/Telephony/Security and Control Systems Helpdesk:****
Telephone: [phone]****
****
Helpdesk Customers: http://myhelp.myitdepartment.net****
Blog: http://blog.myitdepartment.net****
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/****
****
** **
--
Michael Picher, Director of Technical Services
eZuce, Inc.****
300 Brickstone Square****
Suite 201****
Andover, MA. 01810****
O.978-296-1005 X2015
M.207-956-0262
@mpicher <http://twitter.com/mpicher> ****
linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
www.ezuce.com****
** **
------------------------------------------------------------------------------------------------------------
****
There are 10 kinds of people in the world, those who understand binary and
those who don't.****
** **
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

[Kurt Albershardt]

Given my past VoIP experiences (it's been a few years), I just was about to ask why sipX would not support multiple NICs and then it hit me: While multiple NICs make perfect sense in a B2BUA architecture, they really do not for a pure SIP proxy design.

Is sipXbridge still the preferred path for isolating sipX from the outside world and avoiding NAT traversal? More particularly, in a "sipX behind pfSense" configuration is it still a good way to link to the outside world?


thanks~

Post by m***@mattkeys.net
Therein lies your problem. At no time has multiple NIC's ever been supported or recommended.
I would suggest you drop back down to one NIC. The issue you are having WILL reintroduce iteself as the system runs, I guarantee it.

[Tony Graziano]

Sipxbridge is just an option. A properly configured SBC works well too. It
absolutely depends on your requirements.

Post by Kurt Albershardt
Given my past VoIP experiences (it's been a few years), I just was about
to ask why sipX would not support multiple NICs and then it hit me: While
multiple NICs make perfect sense in a B2BUA architecture, they really do
not for a pure SIP proxy design.
Is sipXbridge still the preferred path for isolating sipX from the outside
world and avoiding NAT traversal? More particularly, in a "sipX behind
pfSense" configuration is it still a good way to link to the outside world?
thanks~

Post by Tony Graziano
Therein lies your problem. At no time has multiple NIC's ever been

supported or recommended.

Post by Tony Graziano
I would suggest you drop back down to one NIC. The issue you are having

WILL reintroduce iteself as the system runs, I guarantee it.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

[Kurt Albershardt]

Thanks - think I'll give sipXbridge a whirl.

I notice that http://wiki.sipfoundry.org/display/sipXecs/SIP+Trunking has not been updated since April 2011. Are there changes I should be aware of?

Sipxbridge is just an option. A properly configured SBC works well too. It absolutely depends on your requirements.
Given my past VoIP experiences (it's been a few years), I just was about to ask why sipX would not support multiple NICs and then it hit me: While multiple NICs make perfect sense in a B2BUA architecture, they really do not for a pure SIP proxy design.
Is sipXbridge still the preferred path for isolating sipX from the outside world and avoiding NAT traversal? More particularly, in a "sipX behind pfSense" configuration is it still a good way to link to the outside world?
thanks~

Post by m***@mattkeys.net
Therein lies your problem. At no time has multiple NIC's ever been supported or recommended.
I would suggest you drop back down to one NIC. The issue you are having WILL reintroduce iteself as the system runs, I guarantee it.

[Tony Graziano]

No. It no longer automatically puts "operator" in as the "Incoming Calls
Desitnation", but that does not affect anything other than how to configure
DID's, which people would have to know anyway.

Post by Kurt Albershardt
Thanks - think I'll give sipXbridge a whirl.
I notice that http://wiki.sipfoundry.org/display/sipXecs/SIP+Trunking has
not been updated since April 2011. Are there changes I should be aware of?
Sipxbridge is just an option. A properly configured SBC works well too. It
absolutely depends on your requirements.

Post by Kurt Albershardt
Given my past VoIP experiences (it's been a few years), I just was about
to ask why sipX would not support multiple NICs and then it hit me: While
multiple NICs make perfect sense in a B2BUA architecture, they really do
not for a pure SIP proxy design.
Is sipXbridge still the preferred path for isolating sipX from the
outside world and avoiding NAT traversal? More particularly, in a "sipX
behind pfSense" configuration is it still a good way to link to the outside
world?
thanks~

Post by Tony Graziano
Therein lies your problem. At no time has multiple NIC's ever been

supported or recommended.

Post by Tony Graziano
I would suggest you drop back down to one NIC. The issue you are having

WILL reintroduce iteself as the system runs, I guarantee it.

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