After a migration to asterisk 1.6, I don't receive sip incoming calls anymore.
As fas as I understand the SIP debug traces, my server receives the request and reject it: ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ <--- SIP read from UDP:212.27.52.5:5060 ---> INVITE sip:[email protected]:5060;transport=udp SIP/2.0 Call-ID: [email protected] Contact: <sip:172.17.20.241:5062> Content-Type: application/sdp CSeq: 239836027 INVITE From: "096160XXXX" <sip:[email protected];user=phone>;tag=25151-GA-0eaf098c-32a97dc05 Max-Forwards: 28 Record-Route: <sip:C=on-88.165.134.117.5060;[email protected]:5060;lr> To: <sip:[email protected];user=phone> Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RDKI-00720204-55801b4f Allow: UPDATE,REFER,INFO User-Agent: Cirpack/v4.41c (gw_sip) Content-Length: 173 v=0 o=cp10 125830752022 125830752022 IN IP4 212.27.52.129 s=SIP Call c=IN IP4 212.27.52.129 t=0 0 m=audio 36480 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:30 <-------------> --- (13 headers 9 lines) --- == Using SIP RTP CoS mark 5 Sending to 212.27.52.5 : 5060 (no NAT) Using INVITE request as basis request - [email protected] Found peer 'freephonie_appelsortant' for '096160XXXX' from 212.27.52.5:5060 asterisk*CLI> <--- Reliably Transmitting (no NAT) to 212.27.52.5:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RDKI-00720204-55801b4f;received=212.27.52.5 From: "096160XXXX" <sip:[email protected];user=phone>;tag=25151-GA-0eaf098c-32a97dc05 To: <sip:[email protected];user=phone>;tag=as03dcbe68 Call-ID: [email protected] CSeq: 239836027 INVITE Server: Asterisk PBX 1.6.2.0~rc2-0ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78360854" Content-Length: 0 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Some googling kind of suggest that this might be because for my ISP my username is also my phone number: http://lists.digium.com/pipermail/asterisk-dev/2009-January/036259.html > The problem arises since you use phone numbers as identifiers for the > users. This is not a good thing (TM) and should be avoided. The > dialplan is where you route phone numbers. Devices should have device > names that you address in the dialplan on the extension that is > supposed to connect to one or several devices. Am I right or must I search elsewhere? Whether it's a good thing or not, I doubt I can convince Free (http://free.fr) which is one of the biggest ISPs in France to change their policy so that I can receive SIP calls again... If my diagnostic is right, is there a way to work around this issue with asterisk 1.6? Thanks, Eric -- Eric van der Vlist <[email protected]> Dyomedea (http://dyomedea.com) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
