[SR-Users] Repeated 200 OK from Enswitch

Daniel-Constantin Mierla miconda at gmail.com
Tue May 12 09:45:52 CEST 2015


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 at PROVIDERIP:5080>;tag=as65919d92^M
> To: <sip:PHONENUMBER at PROVIDERIP>;tag=as260fefaa^M
> Call-ID: 271ac7a174d613cd0b94504353733a2c at 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 at 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.. at ..v....g.v.......T.ACK sip:PHONENUMBER at 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 at PROVIDERIP:5080>;tag=as65919d92^M
> To: <sip:PHONENUMBER at PROVIDERIP>;tag=as260fefaa^M
> Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>^M
> Call-ID: 271ac7a174d613cd0b94504353733a2c at PROVIDERIP^M
> CSeq: 103 ACK^M
> User-Agent: Elastix 3.0^M
> Content-Length: 0^M
>
>
>
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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com

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