Closed jama00 closed 1 year ago
Hello,
The log shows that Asterisk does not think it received a response to the INVITE:
[Feb 1 12:47:13] WARNING[115405]: chan_sip.c:4342 retrans_pkt: Retransmission timeout reached on transmission 10bd18dd-03cc000b-151283d7-aef882e5@10.0.101.165 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Feb 1 12:47:13] WARNING[115405]: chan_sip.c:4366 retrans_pkt: Hanging up call 10bd18dd-03cc000b-151283d7-aef882e5@10.0.101.165 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
The phone does send an ACK, but for some reason chan_sip does not match the dialog. You may be able to work-around the issue by setting canreinvite=no
on those peers so Asterisk doesn't have to re-INVITE them to change the RTP destinations.
Hello, this helped, but all RTP is now routed through Asterisk. Is it possible to make directmedia working in this case?
And also interesting, that reinvites works when there is a direct call, without conferencing. At least, swithing to MOH and back does not belongs to breaking call.
And answering own question, canreinvite=update option (or directmedia=update) resolves this issue. RTPs are routed between phones directly and conferences are working too.
Thanks for testing that. I will update the site's documentation.
Good day! I have Asterisk 18.16.0, patched with 18.15 patch, but it applied without any warnings, so i think it's doesnt matter. And three phones, connected to it - Cisco 7965, 7811 and 8851. debian-pbx*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description 1001/1001 10.0.101.165 D No No 50671 Unmonitored T.T. Testov <-- This is Cisco 7965 1002/1002 10.0.101.155 D No No 49768 Unmonitored Cisco 8851 1003/1003 10.0.101.157 D No No 51500 Unmonitored Cisco 7811 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
It has no issues with generic calls, but when i make conference, it is terminating after 32 sec.
in SIP debug (and pcap) there is a 503 Unavailable error, which replied to this INVITE message from phone:
The situation is reproduced when organizing a conference from any of the three phones.
sip_debug_cisco_conf_008.txt conf_cisco_sip_008_sip.zip this is the debug log and tcpdump session of this
Is it some issue with INVITE parsing or something other?