actually, the itsp's explanation is poor. please describe how the call originates. is it coming from outside and then being forwarded? Does it originiate inside and is just going to the ITSP? Can you make normal outbound calls without audio issues and this only happens with forwarded calls?
your pcap is MOST useful at the firewall OR by getting a proper siptrace from sipx. What you would typically find is the issue with audio issues is a RTP port mismatch. So the firewall matters and so does the ITSP. Who is the ITSP? What is the firewall? Is the firewall sip alg/helper off? Is outbound NAT static port? If you ran a pcap and viewed it I suspect you would see sipx sending the audio out on port 305xx and the ITSP sending back to sipx on a random port. On Wed, Nov 23, 2011 at 3:59 PM, Tommy Elliott <[email protected]> wrote: > First , I am running Sipxecs 4.4.0 on CentOS with a 2.6.18 kernel. It is > running in a small office with less than 10 phones being used. **** > > ** ** > > The issue I am having is:**** > > When I forward the phones via the Sipx server to an external phone, the > call is established but there is no audio on either end.**** > > ** ** > > Here are my Hunt Group configurations I was playing with during testing:** > ** > > ** ** > > Hunt Group configuration 1:**** > > 1 – Initially Call User 13**** > > 2 – At the same time User 11**** > > 3 – If no response User 10**** > > 4 – At the same time User 12**** > > ** ** > > Hunt Group configuration 2: > 1 - Initially Call User 13 > 2 - If no response User 11 > 3 - At the same time User 10 > 4 - At the same time User 12 > > Hunt Group configuration 3: > 1 - Initially Call User 12 > 2 - If no response User 13 > 3 - At the same time User 11 > 4 - At the same time User 10**** > > ** ** > > I did three separate tests: > 1) Forwarding User 13’s desk phone to her cell phone using Hunt Group > configuration 1; sip.pcap > 2) Forwarding User 13’s desk phone to her cell phone using Hunt Group > configuration 2; sip1.pcap > 3) Forwarding User 12's desk phone to User 13's cell phone using Hunt > Group configuration 3; sip2.pcap**** > > ** ** > > I installed Wireshark onto the SIPX server to run wireshark on it directly > via the following command: > tshark src net 192.168.10.0\24 -w /root/sip.pcap > > All three tests gave me the same results as normal; the call gets > established but there is no audio on either end. > > Lamen terms about issue (input given by VoIP Gateway Provider): The > carrier is receiving the call and establishing the call but does not know > where to send the audio stream. > > Technical detail about issue (input given by VoIP Gateway Provider): The > PBX is sending a 302 Moved Temporarily but it should be sending a REINVITE. > The PBX is sending an ACK before the call is even setup. This is why it > looks like it is being rejected.**** > > ** ** > > What I am really trying to determine if this is an issue on my end at the > Sipx Server or if this is an issue on the VoIP Gateway Providers. His > response/input is that the Sipx Server is not handling the call correctly. > It should be sending a Reinvite so that the carrier knows where to send the > audio stream back to. I have sent all three *.pcap files over to the > Engineering team of the VoIP Gateway Provider but I wanted to get some > input from the community as well. If the *.pcap files will help find a > resolution or if any other information is needed then please let me know > and I will try to make it available as soon as possible.**** > > ** ** > > Thanks,**** > > Tommy Elliott**** > > Digital Network Solutions, LLC************ > > [email protected]**** > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services!
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
