Mathieu Chouquet-Stringer
2006-Jun-29 01:50 UTC
[Asterisk-Users] Asterisk with Sipbroker calling / routing problem
Hello all, I've been using * for quite some time and yesterday I decided to add sipbroker to my config. It was pretty simple and it works for some numbers (e.g. I can call *258-9123, UK date & time - which is on the "phone numbers you can call" page -) but fails for some others. For example I've got a friend who's at freephonie so to call him, I would dial *759608xxxxxxxx (7596 being freephonie.net). When I do that, I get the following error: Jun 29 10:27:21 NOTICE[7916]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to '<sip:0001@somehost.somedomain.tdl>;tag=as32d2cdfe' And here's a snippet of what I get from 'sip debug': -------------------------------------------------------------------------- <-- SIP read from 24.196.79.163:5060: SIP/2.0 407 authentication required Allow: UPDATE,REFER Call-ID: 73833b4d1ddb389d7a8114e4684f091b@somehost.somedomain.tdl Contact: <sip:212.27.52.5:5060> CSeq: 102 INVITE From: <sip:0001@somehost.somedomain.tdl>;tag=as32d2cdfe Proxy-Authenticate: Digest realm="freephonie.net",nonce="012dd3995b84e8f56ca34a7201a0c6ff",opaque="012daad2220ed2c",stale=false,algorithm=MD5 Record-Route: <sip:24.196.79.163;lr;ftag=as32d2cdfe> Server: Cirpack/v4.40 (gw_sip) To: <sip:*759608xxxxxxxx@sipbroker.com>;tag=01-08146-012dd3ab-3b2383163 Via: SIP/2.0/UDP 172.16.1.1:5060;received=86.216.233.69;rport=5060;branch=z9hG4bK76bd560d Content-Length: 0 --- (12 headers 0 lines)--- Transmitting (no NAT) to 24.196.79.163:5060: ACK sip:*759608xxxxxxxx@sipbroker.com SIP/2.0 Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK76bd560d;rport From: <sip:0001@somehost.somedomain.tdl>;tag=as32d2cdfe To: <sip:*759608xxxxxxxx@sipbroker.com>;tag=01-08146-012dd3ab-3b2383163 Contact: <sip:0001@172.16.1.1> Call-ID: 73833b4d1ddb389d7a8114e4684f091b@somehost.somedomain.tdl CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 29 10:27:21 NOTICE[7916]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to '<sip:0001@somehost.somedomain.tdl>;tag=as32d2cdfe' Transmitting (NAT) to 172.16.1.19:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.16.1.19:5060;branch=z9hG4bK9954d222975cdcc1;received=172.16.1.19 From: <sip:0001@asterisk;user=phone>;tag=2858979361 To: <sip:*759608xxxxxxxx@asterisk;user=phone>;tag=as4eecd6f3 Call-ID: 4074108287@172.16.1.19 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*759608xxxxxxxx@172.16.1.1> Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing -------------------------------------------------------------------------- Here's what I got in sip.conf for sipbroker: [sipbroker-out] type=peer fromuser=0001 fromdomain=somehost.somedomain.tdl host=sipbroker.com port=5060 canreinvite=yes qualify=yes Any idea what's going on? I've been reading quite a few papers about SIP authentication but I still fail to understand what's really happening (or is freephonie not 'open')? Any help is welcome! Cheers, -- Mathieu Chouquet-Stringer ml2news@free.fr