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We use Kamailio 5.5.0 and topos/topos_redis as well as the uac module to register with sipconnect 1.1 based trunks. When we want to originate a call via such a trunk, we encounter the following problem with ACK routing.
200 OK sent by Trunk
U 217.10.68.150:5060 -> 172.31.16.103:5060 #7583 SIP/2.0 200 OK. Via: SIP/2.0/UDP 3.67.244.157:5060;branch=z9hG4bK496f.744a3c48b24be1ba5fdda68cf5fe3878.1. Record-Route: <sip:172.20.40.8;lr>. Record-Route: <sip:217.10.68.150;lr>. Call-ID: 7cd83e052202a97f4e28728276e0c758@35.157.121.101:5060. From: "01705441258" <sip:2908206t0@sipconnect.sipgate.de>;tag=as010d26b4. To: <sip:+493046996025@voip.sandbox.aaron.ai>;tag=221f5413-4c48-4f53-97e8-d96ea6413f26. CSeq: 102 INVITE. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: <sip:217.10.77.61:5060>. Supported: replaces. Content-Type: application/sdp. Content-Length: 242. . v=0. o=- 56188352 56188354 IN IP4 212.9.44.180. s=sGW. c=IN IP4 212.9.44.180. t=0 0. m=audio 27558 RTP/AVP 8 101. a=maxptime:150. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=sendrecv. a=rtcp:27559. a=ptime:20.
ACK sent by Kamailio
ACK sip:217.10.68.150:5060 SIP/2.0. Via: SIP/2.0/UDP 3.67.244.157:5060;branch=z9hG4bK496f.af6a45e3907d2bea8ea65cc42f223fb2.0. Max-Forwards: 69. From: "01705441258" <sip:01705441258@35.157.121.101>;tag=as010d26b4. To: <sip:+493046996025@voip.sandbox.aaron.ai:5060>;tag=221f5413-4c48-4f53-97e8-d96ea6413f26. Call-ID: 7cd83e052202a97f4e28728276e0c758@35.157.121.101:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 13.38.1. Content-Length: 0. Route: <sip:217.10.68.150;lr>. Route: <sip:172.20.40.8;lr>. Contact: <sip:btpsh-61a6122b-a-1@3.67.244.157>. .
The ACK is not accepted by the trunk, because the R-URI does not contain the address fromt the 200 OK/Contact. 200 OK is then retransmitted by the trunk.
kamailio-uac-topos-bug-report.zip
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