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<p class="MsoNormal">I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer’s endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn’t
have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to Asterisk with SDP information for a different RTP port number. Asterisk is ACKing the RE-INVITE, but
never changes media over to the new port number. <o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">AdTran is saying it’s Asterisk’s problem, since the Wireshark trace shows Asterisk is ACKing the re-invite but not changing ports. I do see that the Session ID number is different in the two invites (the REINVITE has a higher ID number
than the original 200 OK that sets up the call – my test call was inbound to the NV7100). However, the REINVITE’s version number is lower (1) than the 200 OK’s SDP version number (which was the same as the SDP Session ID number). I see in the sip.conf.sample
file that “By default, Asterisk will honor the session version number in SDP packets and will only modify the SDP session if the version number changes”. Given that I don’t have ignoresdpversion=yes either globally or for this peer, does this mean that Asterisk
will only honor new SDP packets if the version is higher, or will it honor any change? Or should I be looking somewhere else?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thank you,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Noah Engelberth<o:p></o:p></p>
<p class="MsoNormal">MetaLINK Technologies<o:p></o:p></p>
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