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](https://github.com/kamailio/kamailio/fil…
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