David Siegel
2008-Jun-30 05:26 UTC
[asterisk-users] Asterisk to Broadvoice SIP peer fails in 1.6.9-beta9
In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now, with the same SIP configuration, I cannot establish the peer. I've enclosed a SIP log in the hope that someone can help me analyze this failure. I'd guess the issue is NAT related and wondering if someone can spot a problem in the logs, below. Some details to help read this log (I've changed these numbers for privacy purposes): . My Asterisk server is behind a firewall. It's internal address is 192.168.71.1. . My public IP address is 123.123.123.123 . I am calling 2125551212 . My Broadvoice phone number is 9145551212 Here is the log: == Using SIP RTP CoS mark 5 -- Executing [912125551234 at dialplan-siegel:1] Macro("SIP/siegeld-00e08e00", "dial-sip,12125551234 at sip.\ broadvoice.com") in new stack -- Executing [s at macro-dial-sip:1] Dial("SIP/siegeld-00e08e00", "SIP/12125551234 at sip.broadvoice.com") i\ n new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.71.7 port 18596 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (NAT) to 123.123.123.123:5060: INVITE sip:12125551234 at sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport Max-Forwards: 70 From: "David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4 To: <sip:12125551234 at sip.broadvoice.com> Contact: <sip:9145551234 at 192.168.71.7> Call-ID: 6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-beta9 Date: Mon, 30 Jun 2008 05:13:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 256 v=0 o=root 593017814 593017814 IN IP4 192.168.71.7 s=Asterisk PBX 1.6.0-beta9 c=IN IP4 192.168.71.7 t=0 0 m=audio 18596 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 12125551234 at sip.broadvoice.com stsca1*CLI> <--- SIP read from UDP://123.123.123.123:5060 ---> SIP/2.0 100 Trying Call-ID: 6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com CSeq: 102 INVITE From: "David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4 To: <sip:12125551234 at sip.broadvoice.com> Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd Content-Length: 0 <-------------> --- (7 headers 0 lines) --- stsca1*CLI> <--- SIP read from UDP://123.123.123.123:5060 ---> SIP/2.0 403 Forbidden Call-ID: 6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com CSeq: 102 INVITE From: "David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4 To: <sip:12125551234 at sip.broadvoice.com>;tag=lmno Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd User-Agent: Asterisk PBX 1.6.0-beta9 Content-Type: application/sdp Content-Length: 188 v=0 o=1213832004 593017814 593017814 IN IP4 192.168.71.7 s=- c=IN IP4 192.168.71.7 t=0 0 m=audio 18596 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 <-------------> --- (9 headers 9 lines) --- Transmitting (NAT) to 123.123.123.123:5060: ACK sip:12125551234 at sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport Max-Forwards: 70 From: "David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4 To: <sip:12125551234 at sip.broadvoice.com>;tag=lmno Contact: <sip:9145551234 at 192.168.71.7> Call-ID: 6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0-beta9 Content-Length: 0 --- [Jun 30 01:13:51] WARNING[3023]: chan_sip.c:14738 handle_response_invite: Received response: "Forbidden" f\ rom '"David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4' -- SIP/sip.broadvoice.com-00e0ddb0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s at macro-dial-sip:2] Goto("SIP/siegeld-00e08e00", "s-CONGESTION,1") in new stack -- Goto (macro-dial-sip,s-CONGESTION,1) -- Executing [s-CONGESTION at macro-dial-sip:1] PlayTones("SIP/siegeld-00e08e00", "congestion") in new st\ ack -- Auto fallthrough, channel 'SIP/siegeld-00e08e00' status is 'CONGESTION' Really destroying SIP dialog '6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com' Method: INVI -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080630/9158eb55/attachment.htm