Rather than use an old unsupportable version, produce a pcap from your
firewall or produce a siptrace from sipx itself.
I don't think your off the cuff observation is exactly right on targetm .
Version 4.2 used its own media server while 4.4 uses FreeSWITCH and there
are significant close changes.
You could also indicate whether or not you followed a tutorial on how to
properly configure pfsense and who the itsp is.
On Oct 18, 2012 8:12 PM, "Henry Kwan" <***@yahoo.ca> wrote:
> First of all, allow me to thank everyone who had given me advice on the
> problem that I have been encountering: The problem of the inability to have
> external calls transferred to voice mail when external calls were not
> answered, using version 4.4.0. I am, somewhat, happy that I've resolved
> the problem. The resolution has nothing to do with actually identifying
> the problem but, rather, avoiding it. Let me explain.
>
> All the advices in suggesting that my router, WRVS4400N or the RV016, may
> have been the root of the problem turned out to be wrong. My ITSP was also
> not the source of the problem. I downloaded, installed, used pfSense as my
> firewall/router and the same problem persisted. After much reading and
> searching on the net, I've decided to use 4.2.1 instead of 4.4.0. Viola
> this version worked just fine with the same settings/configuration and same
> hardware, including the routers. All the routers, WRVS4400N, RV016, and
> pfSense, worked.
>
> My obvious conclusion is that one, or more, bug was introduced into 4.4.0
> that caused this behaviour but regression testing of the release did not
> catch it. I am, however, surprised that no one else is reporting this
> behaviour, or bug. Perhaps someone already did but I simply missed it.
>
> It is all good that this problem is out of my way.
>
> I have another observation that I'd like to seek advice. This observation
> is applicable to both 4.2.1 and 4.4.0. I've observed that sometimes after
> making changes to configurations and restarted the required processes, as
> prompted by sipXecs, I could not make external calls but internal calls and
> receiving external calls were just fine. Then I did "Send Profile" to my
> server to restart everything but that was also a hit and miss (meaning
> sometimes the problem of not able to make external calls went away but
> sometime not). I then did "service sipxecs restart" on the command line
> but that was also a hit and miss. This problem was also observed even if
> no configuration changes were made but simply restarting the sipxecs
> processes using methods mentioned above would cause the same observed
> problem. There were no changes on internal and external hardware either.
> So the observed problem had nothing to do with configuration changes. When
> this problem occurred, my phone (SPA942) would show "Calling" then quickly
> show "Forbidden".
>
> My off-the-cuff conclusion is that there must be some race conditions or
> out-of-order events, for lack of a better term, that sipXecs encountered
> but could not consistently resolve or handle properly, leading to this
> condition. I may be totally wrong here but I cannot explain why restarting
> the application without changing any configuration and hardware will cause
> this inconsistent behaviour on the side of the application.
>
> Please excuse my long submission and thank you for your attention.
>
> Best regards,
>
> Henry Kwan
>
>
> ------------------------------
> ***
> From:* Richard Bruce <***@dimensionalcom.com>
> *To:* 'Discussion list for users of sipXecs software' <
> sipx-***@list.sipfoundry.org>
> *Sent:* Wednesday, October 17, 2012 6:00:23 AM
> *Subject:* Re: [sipx-users] External calls cannot be transferred to voice
> mail (sipXecs 4.4.0)
>
> You should probably check the DNS settings on the gateway. I have had
> this problem on multiple analog gateways, having forgotten to set this.
>
>
>
> Richard Bruce
> Dimensional Communications
> 7915 S. Emerson Ave, Suite 131
> Indianapolis, IN 46237
> (317) 215-4199- office
> (317) 946-1899 - cell
> ------------------------------
> *From:* sipx-users-***@list.sipfoundry.org [mailto:
> sipx-users-***@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* Friday, October 12, 2012 10:31 AM
> *To:* Henry Kwan; Sipx-users list
> *Subject:* Re: [sipx-users] External calls cannot be transferred to voice
> mail (sipXecs 4.4.0)
>
> Are you configuring the spa942 manually? If so, do t do that and let sipx
> configure it. Resist the urge to change the configuration for the phone
> within sipx. Explain how you are configured (is sipx DNS and dhcp server),
> etc.
> On Oct 12, 2012 10:27 AM, "Henry Kwan" <***@yahoo.ca> wrote:
> Hi Todd,
>
> Thank you for your response and your assurance that the combination of
> SPA942 and SipXecs 4.4 works.
>
> I am just curious regarding the transfer to voice mail since I am not
> knowledgeable on the sequence of operation. How is the signalling
> different between transfer to voice mail from an internal call and that for
> an external call? Is it correct to say that for an internal call to voice
> mail transfer, only the phone and the SIP server are involved; for an
> external call, the ITSP, SIP server, and phone are involved (therefore the
> router and ITSP may affect this operation)? But the call has already been
> handed to the SIP server, so why does the ITSP need to get into the scene?
> If the ITSP is not involved, what is the difference in handling transfer to
> voice mail between an internal and external call?
>
> I apologize for all these questions but I just am mystified by my
> encounters and observations.
>
> Thanks and best regards,
>
> Henry Kwan
>
> *From:* Todd Hodgen <***@frontier.com>
> *To:* 'Henry Kwan' <***@yahoo.ca>; ' Discussion list for users of
> sipXecs software ' <sipx-***@list.sipfoundry.org>
> *Sent:* Friday, October 12, 2012 12:39:02 AM
> *Subject:* RE: [sipx-users] External calls cannot be transferred to voice
> mail (sipXecs 4.4.0)
>
> Henry, I cant speak to the router, or your ITSP provider. I can
> state that I have a site running on 4.4 with a single server, server
> provides DHCP and DNS, and works with SPA942 phones. I did not use the
> wiki recommendations. I simply provisioned them via the management
> templates and they work perfectly.
>
> Trunks are provided via a PRI gateway Ive used Epygi and Patton
> gateways at this site with great results from both of them.
>
> I would suggest router or ITSP are your issue, as others have.
>
> VOIP.ms is a low cost ITSP provider that for a minimum investment you
> can use to test. We know they work, and for a few bucks you can save
> yourself some time in troubleshooting.
>
> *From:* sipx-users-***@list.sipfoundry.org [mailto:
> sipx-users-***@list.sipfoundry.org] *On Behalf Of *Henry Kwan
> *Sent:* Thursday, October 11, 2012 8:24 PM
> *To:* Tony Graziano
> *Cc:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] External calls cannot be transferred to voice
> mail (sipXecs 4.4.0)
>
> The router, Linksys WRVS4400N, that I am using is not a home router.
> It is a small business router. Having said that it still may not mean it
> is a suitable router for SipX.
>
> I managed to obtain another router and do more testing tonight. The
> router is a Linksys/Cisco RV016. It has one-to-one NAT and I set it up to
> have a one-to-one NAT entry between my internal sipx server and the
> router's external interface.
>
> Using the RV016, the following test results were obtained (please note
> that I had to port forward 5080, and 30000 to 31000, otherwise external
> calls would come through with just one-to-one NAT setup and enabled):
>
> All the previous test results remained exactly the same. That is to say
> internal calls could be transferred to voice mail when no one answer the
> calls but external calls could not.
>
> I then setup forwarding directly to voice mail by calling the external
> voice mail DID number that I setup. That worked!!
>
> I am beginning to think that it may have to do with how the SPA942
> operates or it was not setup properly via the sipxecs web interface. But I
> am not knowledgeable enough to examine and change the settings on the
> SPA942.
>
> If anyone can give me suggestions to troubleshoot this problem, I'd much
> appreciate it.
>
> Best regards,
>
> Henry Kwan
> *From:* Tony Graziano <***@myitdepartment.net>
> *To:* Henry Kwan <***@yahoo.ca>
> *Cc:* Discussion list for users of sipXecs software <
> sipx-***@list.sipfoundry.org>
> *Sent:* Thursday, October 11, 2012 11:35:38 AM
> *Subject:* Re: [sipx-users] External calls cannot be transferred to voice
> mail (sipXecs 4.4.0)
>
> Tested by who? Just because it works as a home router for voip doesn't
> mean it will probably work for your office hosting a PBX, BIG FAT
> difference.
>
> On Thu, Oct 11, 2012 at 12:18 PM, Henry Kwan <***@yahoo.ca> wrote:
> > Do I need one-to-one NAT, or symmetric NAT? I bought this router
> because it
> > had been tested to work with VoIP, whatever that means, but I forgot the
> > source of this information.
> >
> > From: Tony Graziano <***@myitdepartment.net>
> > To: Henry Kwan <***@yahoo.ca>; Discussion list for users of sipXecs
> > software <sipx-***@list.sipfoundry.org>
> > Sent: Thursday, October 11, 2012 9:28:30 AM
> > Subject: Re: [sipx-users] External calls cannot be transferred to voice
> mail
> > (sipXecs 4.4.0)
> >
> > I don't think the router is compatible with the ability to 1:1 NAT or
> > do NAT without changing (randomizing) the source port. I would get
> > thee to a router that will do thusly. Even if you do all of the above,
> > you will likely have frequent or all the time broken audio.
> >
> > On Thu, Oct 11, 2012 at 10:37 AM, Henry Kwan <***@yahoo.ca> wrote:
> >> I am a total newbie on SipXecs. I am also green when it comes to the
> SIP
> >> and VoIP PBX scene. Please excuse my seemingly simple question.
> >>
> >> The problem that I am encountering, essentially, is that external calls
> >> cannot be transferred to voice mail when a call is not answered.
> Internal
> >> calls that were not answered were transferred to voice mail without a
> >> problem.
> >>
> >> My setup:
> >> - SipXecs 4.4.0 installed from the download ISO and updated to the
> latest
> >> patches with yum. OS is also updated to Centos 5.8, with the latest
> >> patches.
> >> - Phones are Linksys SPA942 only, no other phones are on the system.
> Only
> >> 3
> >> phones are on the system.
> >> - Domain: mydomain.company.com. company.com is registerd but
> >> mydomain.company.com is local/internal and the DNS server is the Sipx
> PBX.
> >> - Sipx PBX is the DHCP server, dhcp.conf file is edited to assign a
> >> limited
> >> range of IP addresses. No other dhcp servers are on the subnet.
> >> - The workarounds stated on the sipfoundry wiki for the SPA942 are
> >> implemented, i.e.:
> >> a. MOH Server: ~~mh~@mydomain.company.com
> >> b. Message Waiting: checked
> >> c. Mailbox ID: $USER_ID
> >> d. Voice Mail Server: ***@mydomain.company.com. I have
> >> also changed mydomain.company.com to the IP address of the sipx server.
> >> - Use internal sipXbridge to connect to my SIP trunk. SIP trunk
> >> authenticated successfully and works.
> >> - Router used is Linksys WRVS4400N. Port 5080 and 30000 to 31000 are
> >> forwarded to the SipX PBX.
> >> - Aliases are setup for these 3 phones are set for DID.
> >>
> >> With the above setup, I can dial extensions and have their respective
> >> voice
> >> mail kick-in when a call is not answered. Dial out and DID work as
> well.
> >> The problem that I am encountering now is that voice mail does not
> kick-in
> >> when an external call is not answered. Voice mail does work for
> internal
> >> calls, though.
> >>
> >> I've also added domain aliases of the IP address of the PBX and
> >> PBX.mydomain.company.com <http://pbx.mydomain.company.com/> to the
> setup but that did not help.
> >>
> >> I also setup one of the phones to call forward to another phone, then
> >> voice
> >> mail. The call forwart to another extension worked but call forward to
> >> voice mail did not.
> >>
> >> In desperation, I also added an A record for mydomain.company.com in my
> >> DNS
> >> server but that did not help.
> >>
> >> Lacking the experience of sipX, VoIP PBX, SIP, and network debug tools,
> I
> >> hope experienced SipXecs users can shed some on my plight.
> >>
> >> Thank you.
> >>
> >> Henry Kwan
> >>
> >> _______________________________________________
> >> sipx-users mailing list
> >> sipx-***@list.sipfoundry.org
> >> 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!
> >
> > --
> > 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/
> >
> >
>
>
>
> --
> ~~~~~~~~~~~~~~~~~~
> 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!
>
> --
> 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/
>
>
> _______________________________________________
> sipx-users mailing list
> sipx-***@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: ***@voice.myitdepartment. net<***@voice.myitdepartment.net>
>
> Helpdesk Customers: http://myhelp.myitdepartment. net<http://myhelp.myitdepartment.net/>
> Blog: http://blog.myitdepartment.net
>
> _______________________________________________
> sipx-users mailing list
> sipx-***@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
> _______________________________________________
> sipx-users mailing list
> sipx-***@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
--
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