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]
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