I am helping a friend on one of his sip trunk and couldn't find the way to resolve his problem.
His asterisk's problem is like this: 0. When incoming call to one of his sip trunk, Asterisk reply with "488 Not acceptable here". So the call get dropped. 1. Recently upgraded Elastix with Asterisk 1.4.33 2. Was working fine before the upgrade 3. There are total 4 SIP trunks connected to different providers. All others works fine. 4. All codecs are allowed. 5. I setup his account on my Asterisk as a SIP trunk, both incoming and outgoing call work fine. (So it is not his provider's problem) 6. I checked his FreePBX style multi sip*.conf files and all seem correct. So what can I do to find out where went wrong on this sip trunk? Thanks. Jian Hers is the debug out put: ============================ <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;rport Via: SIP/2.0/UDP 208.65.xxx.xxx:5061;branch=z9hG4bK-pcerhxpz5hr4addh;rport=5061 Max-Forwards: 69 Record-Route: <sip:208.65.xxx.xxx;lr> Contact: "Anonymous"<sip:208.65.xxx.xxx:5061> To: <sip:[email protected]:5060> From: "CID NAME"<sip:[email protected]:5060>;tag=kvspovbxperbwmfk.o Call-ID: [email protected]~o CSeq: 493 INVITE Expires: 300 Content-Disposition: session Content-Type: application/sdp User-Agent: Sippy cisco-GUID: 4084071434-3712422367-2859401243-560159692 h323-conf-id: 4084071434-3712422367-2859401243-560159692 Content-Length: 109 v=0 o=Sippy 153068680 0 IN IP4 74.205.xxx.xxx s=- t=0 0 m=audio 34772 RTP/AVP 0 c=IN IP4 74.205.xxx.xxx <-------------> --- (17 headers 6 lines) --- Sending to 208.65.xxx.xxx : 5060 (NAT) Using INVITE request as basis request - [email protected]~o Found peer 'vsp06' Found RTP audio format 0 <--- Reliably Transmitting (NAT) to 208.65.xxx.xxx:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;received=208.65.xxx.xxx;rport=5060 Via: SIP/2.0/UDP 208.65.xxx.xxx:5061;branch=z9hG4bK-pcerhxpz5hr4addh;rport=5061 From: "CID NAME"<sip:[email protected]:5060>;tag=kvspovbxperbwmfk.o To: <sip:[email protected]:5060>;tag=as40501684 Call-ID: [email protected]~o CSeq: 493 INVITE User-Agent: 000e082e83c7Linksys/SPA2102-5.2.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]~o' in 6400 ms (Method: INVITE) <--- SIP read from 208.65.xxx.xxx:5060 ---> ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;rport Max-Forwards: 70 To: <sip:[email protected]:5060>;tag=as40501684 From: "CID NAME"<sip:[email protected]:5060>;tag=kvspovbxperbwmfk.o Call-ID: [email protected]~o CSeq: 493 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '[email protected]~o' Method: ACK Really destroying SIP dialog '[email protected]' Method: REGISTER astpbx*CLI> sip set debug off SIP Debugging Disabled -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
