Hello,
can you show both received 200ok + ACK as well as those sent out? It is
important to see how Record-/Route, Contact and r-uri change on the way
to spot where the issue is.
Cheers,
Daniel
On 12/05/15 05:56, Darren Campbell (Primar) wrote:
Hi all
Experiencing a commonly reported issue where calls drop out after 30
seconds or so. Mainly because the provider hangs up after not
recognising/receiving ACK in response to 200 OK.
Unfortunately (or maybe fortunately), I haven't had much experience
with Enswitch so was hoping someone in the community might help guide
me as to which rules Enswitch might be using to match ACKs to calls in
progress. Maybe there is another avenue I should be investigating.
Here's a sample of the 200 OK and ACK that repeats.
13:44:04.155646 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1058
E..>.M..?..Ug.v..........*J.SIP/2.0 200 OK^M
Via: SIP/2.0/UDP
172.21.0.226;rport=5060;branch=z9hG4bKfe94.efbf7fbcaf8bd15243a61fdc9d6d1e78.0^M
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK65f00a0c;rport=5080^M
Record-Route: <sip:PROVIDERIP;lr=on>^M
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes>^M
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes>^M
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as65919d92^M
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as260fefaa^M
Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M
CSeq: 103 INVITE^M
Server: Enswitch^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH^M
Supported: replaces^M
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060>^M
Content-Type: application/sdp^M
Content-Length: 286^M
^M
v=0^M
o=root 2110894460 2110894461 IN IP4 PROVIDERMEDIAIP^M
s=Asterisk PBX 11.3.0^M
c=IN IP4 PROVIDERMEDIAIP^M
t=0 0^M
m=audio 15594 RTP/AVP 0 8 3 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:3 GSM/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=sendrecv^M
13:44:04.164519 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)!A..@..v....g.v.......T.ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0^M
Via: SIP/2.0/UDP
172.21.0.226;branch=z9hG4bKfe94.472e9fc0479de79b4f176cc9585d8880.0^M
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK752b5264;rport=5080^M
Route: <sip:PROVIDERIP;lr=on>^M
Max-Forwards: 69^M
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as65919d92^M
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as260fefaa^M
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>^M
Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M
CSeq: 103 ACK^M
User-Agent: Elastix 3.0^M
Content-Length: 0^M
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