Discussion:
Is sipxProxy using wrong IP address? Causing one-way audio.
Sven Evensen
2012-08-28 11:25:09 UTC
Permalink
We have one system (4.4 in EC2) which uses sipxBridge and an other similar
one which uses unmanaged gateway.

Incoming INVITE to these two system show different behavior when the INVITE
goes from sipxProxy and further in to the system where the UA is really our
server on port 5560. Our server relays calls out to mobile phones.

The first INVITE below is with sipxbridge (sdp removed) shows how the
INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also uses
external IP of sipx. This propagates into the system and causes problems on
the final ACK as seen on in attached pcap in line 76, this I believe stops
media from being relayed.

*1871 19.774448 10.7.7.30 54532 10.7.7.30 5070 SIP/SDP Request: INVITE
sip:***@ec2-50-18-193-48.us-west-1.compute.amazonaws.com*
INVITE sip:***@ec2-50-18-193-48.us-west-1.compute.amazonaws.comSIP/2.0
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Call-Id: bee104006623-503c84f6-2cae7b74-1d175788-***@127.0.0.1-0
Cseq: 17965 INVITE
From: <sip:***@194.145.191.131>;tag=787762526
To: <sip:***@ec2-50-18-193-48.us-west-1.compute.amazonaws.com>
Via: SIP/2.0/TCP 10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
References: bee104006623-503c84f6-2cae7b74-1d175788-***@127.0.0.1
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Contact: <sip:***@10.7.7.30:5090;x-sipX-nonat>
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true

*1382 9.460933 10.5.5.10 37326 10.5.5.10 5070 SIP/SDP Request: INVITE
sip:***@176.34.141.24,*
INVITE sip:***@176.34.141.24 SIP/2.0
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
From: "+442083994301" <sip:+***@94.231.97.9>;tag=as2ba8402f
To: <sip:***@176.34.141.24>
Contact: <sip:+***@94.231.97.9;x-sipX-nonat>
Call-Id: ***@94.231.97.9
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
--
*Sven Evensen, Operations Consultant*

*OnRelay*

Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123 │
mailto:***@onrelay.com <***@onrelay.com> │ www.onrelay.com


This electronic message transmission contains information from OnRelay,
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Tony Graziano
2012-08-28 12:14:55 UTC
Permalink
This might seem off target, but help me understand how the UA is
communicating with the proxy...

First, I have to say the eyebeam UA is VERY old and trying to troubleshoot
against it is problematic to say the least.

Can you provide a sip trace with a single call instead of pcap? At the ITSP
the port is 18402, at the UA it is 30498. I think it would be easier to
follow and understand with a sip trace in this instance.
Post by Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other similar
one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when the
INVITE goes from sipxProxy and further in to the system where the UA is
really our server on port 5560. Our server relays calls out to mobile
phones.
The first INVITE below is with sipxbridge (sdp removed) shows how the
INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also uses
external IP of sipx. This propagates into the system and causes problems on
the final ACK as seen on in attached pcap in line 76, this I believe stops
media from being relayed.
*1871 19.774448 10.7.7.30 54532 10.7.7.30 5070 SIP/SDP Request: INVITE
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP 10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
*1382 9.460933 10.5.5.10 37326 10.5.5.10 5070 SIP/SDP Request: INVITE
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
--
*Sven Evensen, Operations Consultant*
*OnRelay*
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123
www.onrelay.com
This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
delete this message, along with any attachments, from your computer.
Registered in England No 04006093 Š Registered Office 1st Floor, 236 Gray's
Inn Road, London WC1X 8HB
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
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Telephone: 434.984.8430
sip: ***@voice.myitdepartment.net
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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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>
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: ***@voice.myitdepartment.net

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
Joegen Baclor
2012-08-28 12:37:19 UTC
Permalink
The answer is staring us right on the face Tony. The INVITE is towards
sip:***@176.34.141.24 <mailto:sip%***@176.34.141.24> and not
the domain. The proxy thinks it's routing out so it uses the external
IP. Configure the calling UA to use domain.
Post by Tony Graziano
This might seem off target, but help me understand how the UA is
communicating with the proxy...
First, I have to say the eyebeam UA is VERY old and trying to
troubleshoot against it is problematic to say the least.
Can you provide a sip trace with a single call instead of pcap? At the
ITSP the port is 18402, at the UA it is 30498. I think it would be
easier to follow and understand with a sip trace in this instance.
On Tue, Aug 28, 2012 at 7:25 AM, Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other
similar one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when
the INVITE goes from sipxProxy and further in to the system where
the UA is really our server on port 5560. Our server relays calls
out to mobile phones.
The first INVITE below is with sipxbridge (sdp removed) shows how
the INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA
also uses external IP of sipx. This propagates into the system and
causes problems on the final ACK as seen on in attached pcap in
line 76, this I believe stops media from being relayed.
*187119.77444810.7.7.305453210.7.7.305070SIP/SDPRequest: INVITE
INVITE
SIP/2.0
Record-Route: <sip:10.7.7.30:5060;lr>
<sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP
10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP
10.7.7.30:5090;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
*13829.46093310.5.5.103732610.5.5.105070SIP/SDPRequest: INVITE
Record-Route: <sip:176.34.141.24:5060;lr>
<sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070;transport=tcp;lr>
Via: SIP/2.0/TCP
176.34.141.24:5060;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
--
*Sven Evensen, Operations Consultant*
*OnRelay*
Elizabeth House ? 39 York Road, London SE1 7NQ, UK ? +44 (0) 207
<http://www.onrelay.com/>
This electronic message transmission contains information from
OnRelay, Ltd., that may be confidential or privileged. The
information is intended solely for the recipient and use by any
other party is not authorised. If you are not the intended
recipient, be aware that any disclosure, copying, distribution or
use of the contents of this information or any attachment, is
prohibited. If you have received this electronic transmission in
error, please notify us immediately by electronic mail
message, along with any attachments, from your computer.
Registered in England No 04006093 Š Registered Office 1st Floor,
236 Gray's Inn Road, London WC1X 8HB
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
Telephone: 434.984.8426
Helpdesk Customers: http://myhelp.myitdepartment.net
<http://myhelp.myitdepartment.net>
Blog: http://blog.myitdepartment.net
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
Tony Graziano
2012-08-28 13:03:11 UTC
Permalink
I also recall some strange configuration behavior issues with eyebeam, so I
lift an eyebrow when I see it still being used. Confirm how the
registration line looks like for the eyebeam in sipx please?
Post by Joegen Baclor
The answer is staring us right on the face Tony. The INVITE is towards
routing out so it uses the external IP. Configure the calling UA to use
domain.
This might seem off target, but help me understand how the UA is
communicating with the proxy...
First, I have to say the eyebeam UA is VERY old and trying to
troubleshoot against it is problematic to say the least.
Can you provide a sip trace with a single call instead of pcap? At the
ITSP the port is 18402, at the UA it is 30498. I think it would be easier
to follow and understand with a sip trace in this instance.
Post by Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other
similar one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when the
INVITE goes from sipxProxy and further in to the system where the UA is
really our server on port 5560. Our server relays calls out to mobile
phones.
The first INVITE below is with sipxbridge (sdp removed) shows how the
INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also uses
external IP of sipx. This propagates into the system and causes problems on
the final ACK as seen on in attached pcap in line 76, this I believe stops
media from being relayed.
*1871 19.774448 10.7.7.30 54532 10.7.7.30 5070 SIP/SDP Request: INVITE
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP 10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
*1382 9.460933 10.5.5.10 37326 10.5.5.10 5070 SIP/SDP Request: INVITE
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
--
*Sven Evensen, Operations Consultant*
*OnRelay*
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123
www.onrelay.com
This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
delete this message, along with any attachments, from your computer.
Registered in England No 04006093 Š Registered Office 1st Floor, 236 Gray's
Inn Road, London WC1X 8HB
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
Telephone: 434.984.8426
Helpdesk Customers: http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
Blog: http://blog.myitdepartment.net
_______________________________________________
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: ***@voice.myitdepartment.net
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: ***@voice.myitdepartment.net

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
Levend Sayar
2012-08-28 13:18:44 UTC
Permalink
Or add 176.34.141.24 as domain alias ?

_lvnd_
(^_^)
Post by Joegen Baclor
The answer is staring us right on the face Tony. The INVITE is towards
routing out so it uses the external IP. Configure the calling UA to use
domain.
This might seem off target, but help me understand how the UA is
communicating with the proxy...
First, I have to say the eyebeam UA is VERY old and trying to
troubleshoot against it is problematic to say the least.
Can you provide a sip trace with a single call instead of pcap? At the
ITSP the port is 18402, at the UA it is 30498. I think it would be easier
to follow and understand with a sip trace in this instance.
Post by Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other
similar one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when the
INVITE goes from sipxProxy and further in to the system where the UA is
really our server on port 5560. Our server relays calls out to mobile
phones.
The first INVITE below is with sipxbridge (sdp removed) shows how the
INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also uses
external IP of sipx. This propagates into the system and causes problems on
the final ACK as seen on in attached pcap in line 76, this I believe stops
media from being relayed.
*1871 19.774448 10.7.7.30 54532 10.7.7.30 5070 SIP/SDP Request: INVITE
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP 10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
*1382 9.460933 10.5.5.10 37326 10.5.5.10 5070 SIP/SDP Request: INVITE
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
--
*Sven Evensen, Operations Consultant*
*OnRelay*
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123│
This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
delete this message, along with any attachments, from your computer.
Registered in England No 04006093 Š Registered Office 1st Floor, 236 Gray's
Inn Road, London WC1X 8HB
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
Telephone: 434.984.8426
Helpdesk Customers: http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
Blog: http://blog.myitdepartment.net
_______________________________________________
List Archive: http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
Tony Graziano
2012-08-28 13:48:04 UTC
Permalink
The registration should be by sip domain.
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sip: ***@voice.myitdepartment.net
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Post by Levend Sayar
Or add 176.34.141.24 as domain alias ?
_lvnd_
(^_^)
Post by Joegen Baclor
The answer is staring us right on the face Tony. The INVITE is towards
routing out so it uses the external IP. Configure the calling UA to use
domain.
This might seem off target, but help me understand how the UA is
communicating with the proxy...
First, I have to say the eyebeam UA is VERY old and trying to
troubleshoot against it is problematic to say the least.
Can you provide a sip trace with a single call instead of pcap? At the
ITSP the port is 18402, at the UA it is 30498. I think it would be easier
to follow and understand with a sip trace in this instance.
Post by Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other
similar one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when the
INVITE goes from sipxProxy and further in to the system where the UA is
really our server on port 5560. Our server relays calls out to mobile
phones.
The first INVITE below is with sipxbridge (sdp removed) shows how the
INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also
uses external IP of sipx. This propagates into the system and causes
problems on the final ACK as seen on in attached pcap in line 76, this
I believe stops media from being relayed.
*1871 19.774448 10.7.7.30 54532 10.7.7.30 5070 SIP/SDP Request: INVITE
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP 10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
*1382 9.460933 10.5.5.10 37326 10.5.5.10 5070 SIP/SDP Request: INVITE
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
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*Sven Evensen, Operations Consultant*
*OnRelay*
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www.onrelay.com
This electronic message transmission contains information from OnRelay,
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Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~
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Sven Evensen
2012-08-28 18:17:43 UTC
Permalink
In this scenario there is no xlite or eyeeam, just a phantom sipx user. I
believe what Joegen said aout the SIP trunk sending INVITE to ip address
and not to domain might be the answer. I have asked the ITSP
to change their setup, we will see if that helps!
Post by Tony Graziano
The registration should be by sip domain.
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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!
Post by Levend Sayar
Or add 176.34.141.24 as domain alias ?
_lvnd_
(^_^)
Post by Joegen Baclor
The answer is staring us right on the face Tony. The INVITE is
thinks it's routing out so it uses the external IP. Configure the
calling UA to use domain.
This might seem off target, but help me understand how the UA is
communicating with the proxy...
First, I have to say the eyebeam UA is VERY old and trying to
troubleshoot against it is problematic to say the least.
Can you provide a sip trace with a single call instead of pcap? At the
ITSP the port is 18402, at the UA it is 30498. I think it would be easier
to follow and understand with a sip trace in this instance.
Post by Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other
similar one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when the
INVITE goes from sipxProxy and further in to the system where the UA is
really our server on port 5560. Our server relays calls out to mobile
phones.
The first INVITE below is with sipxbridge (sdp removed) shows how the
INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also
uses external IP of sipx. This propagates into the system and causes
problems on the final ACK as seen on in attached pcap in line 76, this
I believe stops media from being relayed.
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP
10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
*1382 9.460933 10.5.5.10 37326 10.5.5.10 5070 SIP/SDP Request: INVITE
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
--
*Sven Evensen, Operations Consultant*
*OnRelay*
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902
www.onrelay.com
This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
delete this message, along with any attachments, from your computer.
Registered in England No 04006093 Š Registered Office 1st Floor, 236 Gray's
Inn Road, London WC1X 8HB
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
Telephone: 434.984.8426
Helpdesk Customers: http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
Blog: http://blog.myitdepartment.net
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*OnRelay*

Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123 │
mailto:***@onrelay.com <***@onrelay.com> │ www.onrelay.com


This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
please notify us immediately by electronic mail (***@onrelay.com) and
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Tony Graziano
2012-08-28 18:32:49 UTC
Permalink
Really? I see in the stuff you posted on this one had a UA as "eyebeam". I
looked at it many times, I still see it. I think maybe there are multiple
devices registered to that subscriber and one is an EYEBEAM. Did you chase
the registration(s) down in sipxconfig?
Post by Sven Evensen
In this scenario there is no xlite or eyeeam, just a phantom sipx user. I
believe what Joegen said aout the SIP trunk sending INVITE to ip address
and not to domain might be the answer. I have asked the ITSP
to change their setup, we will see if that helps!
On Tue, Aug 28, 2012 at 2:48 PM, Tony Graziano <
Post by Tony Graziano
The registration should be by sip domain.
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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!
Post by Levend Sayar
Or add 176.34.141.24 as domain alias ?
_lvnd_
(^_^)
Post by Joegen Baclor
The answer is staring us right on the face Tony. The INVITE is
thinks it's routing out so it uses the external IP. Configure the
calling UA to use domain.
This might seem off target, but help me understand how the UA is
communicating with the proxy...
First, I have to say the eyebeam UA is VERY old and trying to
troubleshoot against it is problematic to say the least.
Can you provide a sip trace with a single call instead of pcap? At
the ITSP the port is 18402, at the UA it is 30498. I think it would be
easier to follow and understand with a sip trace in this instance.
Post by Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other
similar one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when the
INVITE goes from sipxProxy and further in to the system where the UA is
really our server on port 5560. Our server relays calls out to mobile
phones.
The first INVITE below is with sipxbridge (sdp removed) shows how
the INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also
uses external IP of sipx. This propagates into the system and causes
problems on the final ACK as seen on in attached pcap in line 76, this
I believe stops media from being relayed.
*
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP
10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
--
*Sven Evensen, Operations Consultant*
*OnRelay*
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902
www.onrelay.com
This electronic message transmission contains information from
OnRelay, Ltd., that may be confidential or privileged. The information is
intended solely for the recipient and use by any other party is not
authorised. If you are not the intended recipient, be aware that any
disclosure, copying, distribution or use of the contents of this
information or any attachment, is prohibited. If you have received this
electronic transmission in error, please notify us immediately by
with any attachments, from your computer. Registered in England No 04006093
Š Registered Office 1st Floor, 236 Gray's Inn Road, London WC1X 8HB
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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
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<http://sipxcolab2013.eventbrite.com/?discount=tony2013>
Telephone: 434.984.8426
Helpdesk Customers: http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
Blog: http://blog.myitdepartment.net
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*OnRelay*
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123
www.onrelay.com
This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
delete this message, along with any attachments, from your computer.
Registered in England No 04006093 Š Registered Office 1st Floor, 236 Gray's
Inn Road, London WC1X 8HB
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: ***@voice.myitdepartment.net
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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>
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: ***@voice.myitdepartment.net

Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
Sven Evensen
2012-08-28 12:36:12 UTC
Permalink
Tony, sorry I did not explain myself well, there is no xlite in this case

1- Leg 1 call comes in through "unmanaged gw" to a sipx user (no phone
connected).
2- Invite propagates to our server listening at port 5560
3- Our server sends leg 2 Invite out through "unmanaged gw"
4- 200OK propagates from leg 2 back to leg 1 via 5560
5-ACK propagates back from leg 1 to leg 2. This should open for media
relay, but it does not happen do to some IP address mix up. See line 6052

siptrace for leg 1 and 2 plus pcap attached.

This works fine if call comes via 5080, but not when it comes straight to
5060.

Sven
Post by Tony Graziano
This might seem off target, but help me understand how the UA is
communicating with the proxy...
First, I have to say the eyebeam UA is VERY old and trying to troubleshoot
against it is problematic to say the least.
Can you provide a sip trace with a single call instead of pcap? At the
ITSP the port is 18402, at the UA it is 30498. I think it would be easier
to follow and understand with a sip trace in this instance.
Post by Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other
similar one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when the
INVITE goes from sipxProxy and further in to the system where the UA is
really our server on port 5560. Our server relays calls out to mobile
phones.
The first INVITE below is with sipxbridge (sdp removed) shows how the
INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also uses
external IP of sipx. This propagates into the system and causes problems on
the final ACK as seen on in attached pcap in line 76, this I believe stops
media from being relayed.
*1871 19.774448 10.7.7.30 54532 10.7.7.30 5070 SIP/SDP Request: INVITE
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP 10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
*1382 9.460933 10.5.5.10 37326 10.5.5.10 5070 SIP/SDP Request: INVITE
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
--
*Sven Evensen, Operations Consultant*
*OnRelay*
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123│
This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
delete this message, along with any attachments, from your computer.
Registered in England No 04006093 Š Registered Office 1st Floor, 236 Gray's
Inn Road, London WC1X 8HB
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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Tony Graziano
2012-08-28 12:53:58 UTC
Permalink
I only said eyebeam because i saw that in the UA type, I didn't say xlite.
I think I would question how the UA (eyebeam) is configured. Personally I
would use something up to date.
Post by Sven Evensen
Tony, sorry I did not explain myself well, there is no xlite in this case
1- Leg 1 call comes in through "unmanaged gw" to a sipx user (no phone
connected).
2- Invite propagates to our server listening at port 5560
3- Our server sends leg 2 Invite out through "unmanaged gw"
4- 200OK propagates from leg 2 back to leg 1 via 5560
5-ACK propagates back from leg 1 to leg 2. This should open for media
relay, but it does not happen do to some IP address mix up. See line 6052
siptrace for leg 1 and 2 plus pcap attached.
This works fine if call comes via 5080, but not when it comes straight to
5060.
Sven
On Tue, Aug 28, 2012 at 1:14 PM, Tony Graziano <
Post by Tony Graziano
This might seem off target, but help me understand how the UA is
communicating with the proxy...
First, I have to say the eyebeam UA is VERY old and trying to
troubleshoot against it is problematic to say the least.
Can you provide a sip trace with a single call instead of pcap? At the
ITSP the port is 18402, at the UA it is 30498. I think it would be easier
to follow and understand with a sip trace in this instance.
Post by Sven Evensen
We have one system (4.4 in EC2) which uses sipxBridge and an other
similar one which uses unmanaged gateway.
Incoming INVITE to these two system show different behavior when the
INVITE goes from sipxProxy and further in to the system where the UA is
really our server on port 5560. Our server relays calls out to mobile
phones.
The first INVITE below is with sipxbridge (sdp removed) shows how the
INVITE uses the FQDN and the VIA uses internal IP of sipx
The second shows the INVITE use external IP of sipx and the VIA also
uses external IP of sipx. This propagates into the system and causes
problems on the final ACK as seen on in attached pcap in line 76, this
I believe stops media from being relayed.
*1871 19.774448 10.7.7.30 54532 10.7.7.30 5070 SIP/SDP Request: INVITE
Record-Route: <sip:10.7.7.30:5060;lr>
Route: <sip:ec2-50-18-193-48.us-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Cseq: 17965 INVITE
Via: SIP/2.0/TCP 10.7.7.30;branch=z9hG4bK-XX-2aaeiUlgRYKle1rNjQIwZC5nnw
Via: SIP/2.0/UDP 10.7.7.30:5090
;branch=z9hG4bK246b613edf92e02fcf8bed5fda63e0f33533;sipxecs-id=41a42656
Max-Forwards: 20
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
;rel=chain;sipxecs-tag=request-invite-z9hg4bk312.e7b1a027.0
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 179
Date: Tue, 28 Aug 2012 08:44:38 GMT
X-Sipx-Spiral: true
*1382 9.460933 10.5.5.10 37326 10.5.5.10 5070 SIP/SDP Request: INVITE
Record-Route: <sip:176.34.141.24:5060;lr>
Route: <sip:ec2-176-34-141-24.eu-west-1.compute.amazonaws.com:5070
;transport=tcp;lr>
Via: SIP/2.0/TCP 176.34.141.24:5060
;branch=z9hG4bK-XX-0182be3pQ05krfO`9nTh52Amwg
Via: SIP/2.0/UDP 94.231.97.9:5060;branch=z9hG4bK72c9e972;rport=5060
Cseq: 102 INVITE
User-Agent: CME VoIP PBX
Max-Forwards: 20
Date: Tue, 28 Aug 2012 08:24:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
X-Sipx-Spiral: true
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*Sven Evensen, Operations Consultant*
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