Tobias Wolf a écrit : > Johan Dindaine - Asterisk schrieb: > >> Hi every all, >> since a few weeks I came back to asterisk and tried to install version 1.6. >> The installation went fine so I decided to buy new dids on Voxbone. >> >> I have added the sip peers of Voxbone Belgium1 like this in the sip.conf >> [81.201.82.39] >> host=dynamic >> type=friend >> insecure=very >> context=your_context >> canreinvite=no >> qualify=no >> deny=0.0.0.0/0.0.0.0 >> permit=81.201.82.39/255.255.255.255 >> >> but unfortunately when I receive a call I got this nice error: >> handle_request_invite: Failed to authenticate user "075XXXXXXXX" >> <sip:[email protected]>;tag=76596. >> >> I am in doubt now because with the insecure=very, I must receive any >> incoming calls from from voxbone (81.201.82.39) without any problems. >> >> Do you know how to fix this please? >> > > Hi, > > we are also using Voxbone Dids and we have no problems: > > Here is a sample defintion from my sip.conf: > > [81.201.83.14] > host = 81.201.83.14 > type = friend > insecure = port,invite > context = voxbone > canreinvite=no > > Hope this helps ... > > Regards > > Tobias Wolf > > > _______________________________________________ > -- 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 > I just modify my sip.conf file to match your configuration provided above and I also printed the debug that I received from voxbone from a SIP SET DEBUG that you can see below. What I don't get is with insecure=very or insecure=port,invite the IP Address of Voxbone should be able to send me an INVITE request without any problems. I simply don't get it. This is the log that I get for anyone who could help me. Thanks for the help
<--- SIP read from 81.201.82.39:5060 ---> INVITE sip:[email protected] SIP/2.0 Call-ID: [email protected] CSeq: 102 INVITE From: "075054XXXXX" <sip:[email protected]>;tag=7419 To: <sip:[email protected]> Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKfe02774c6b7669b771be486f22b7bad9 Max-Forwards: 69 Content-Type: application/sdp Contact: <sip:[email protected]:5060;transport=udp> User-Agent: Vox Callcontrol Content-Length: 311 v=0 o=root 11023 11023 IN IP4 81.201.82.27 s=session c=IN IP4 81.201.82.27 t=0 0 m=audio 17574 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (11 headers 15 lines) --- Sending to 81.201.82.39 : 5060 (no NAT) Using INVITE request as basis request - [email protected] Found no matching peer or user for '81.201.82.39:5060' [Feb 10 22:25:21] NOTICE[4313]: chan_sip.c:14422 handle_request_invite: Failed to authenticate user "075054XXXXX" <sip:[email protected]>;tag=7419 <--- Reliably Transmitting (no NAT) to 81.201.82.39:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKfe02774c6b7669b771be486f22b7bad9;received=81.201.82.39 From: "075054XXXXX" <sip:[email protected]>;tag=7419 To: <sip:[email protected]>;tag=as082cd51d Call-ID: [email protected] CSeq: 102 INVITE User-Agent: XIVO PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) barthez*CLI> <--- SIP read from 81.201.82.39:5060 ---> ACK sip:[email protected] SIP/2.0 Call-ID: [email protected] CSeq: 102 ACK From: "075054XXXXX" <sip:[email protected]>;tag=7419 To: <sip:[email protected]>;tag=as082cd51d Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKfe02774c6b7669b771be486f22b7bad9 Max-Forwards: 69 User-Agent: Vox Callcontrol Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: ACK barthez*CLI> <--- SIP read from 81.106.106.35:8022 ---> _______________________________________________ -- 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
