I got this working if anyone out there is looking to do the same. See: http://www.dslreports.com/forum/remark,12899866~mode=flat#12899866
After some more experimenting, I discovered that you MUST use the long register statement ala Broadvoice. Unlike Broadvoice the service has been ROCK SOLID. Too bad you must have a regular account first :( On Thu, 17 Feb 2005 19:03:39 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote: > I can't seem to dial out with Voicepulse Open Access service using *. > Incoming works fine. Another user posted a few weeks back that they > were having problems and there are some threads at dslreports.com > about this as well. Maybe someone here can figure out what the issue > is from the sip debug info below. I am at a loss. > > The audible error message from Allison is 0984 (from VP server) > > Here is all the pertinent info: > > [sip.conf] > > [general] > port = 5060 > bindaddr = 0.0.0.0 > srvlookup=yes > tos=lowdelay > maxexpirey=3600 > disallow=all > allow=ulaw > musicclass=default > language=en > relaxdtmf=yes > ;useragent=Asterisk PBX > ;nat=yes > > register => s00******:[EMAIL PROTECTED] > > externip=asterisk.briandingman.com > localnet=192.168.1.0/255.255.0.0 > > [voicepulse] > type=friend > context=voicepulse-incoming > username=s00****** > secret=******** > host=access1.voicepulse.com > dtmf=inband > nat=yes > qualify=yes > canreinvite=no > insecure=very > > [1000] > type=friend > host=dynamic > ;callerid=Brian <1000> > dtmfmode=rfc2833 > mailbox=1000 > context=Home > ;nat=no > ;qualify=yes > secret=******** > > Error message from CLI: > -- Executing Macro("SIP/1000-fbdb", "vp-dial|16109951010") in new stack > -- Executing Dial("SIP/1000-fbdb", "SIP/[EMAIL PROTECTED]") in new stack > -- Called [EMAIL PROTECTED] > -- SIP/voicepulse-e009 is making progress passing it to SIP/1000-fbdb > Feb 17 17:08:42 WARNING[8523]: chan_sip.c:6811 handle_response: > Forbidden - wrong password on authentication for INVITE to '"1000" > <sip:[EMAIL PROTECTED]>;tag=as3e632d2a' > -- SIP/voicepulse-e009 is circuit-busy > == Everyone is busy/congested at this time > -- Executing Hangup("SIP/1000-fbdb", "") in new stack > == Spawn extension (macro-vp-dial, s, 2) exited non-zero on > 'SIP/1000-fbdb' in macro 'vp-dial' > == Spawn extension (Home, 16109951010, 1) exited non-zero on 'SIP/1000-fbdb' > -- Got SIP response 481 "Call Leg Does Not Exist" back from 66.234.228.159 > > (Sorry for the length) > SIP Debug info: > > -- Executing Macro("SIP/1000-cd47", "vp-dial|16109951010") in new stack > -- Executing Dial("SIP/1000-cd47", "SIP/[EMAIL PROTECTED]") in new stack > We're at 68.163.52.50 port 15640 > Answering/Requesting with root capability 0x4 (ulaw) > Answering with non-codec capability 0x1 (telephone-event) > 12 headers, 10 lines > Reliably Transmitting: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport > From: "1000" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Date: Thu, 17 Feb 2005 22:10:02 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Type: application/sdp > Content-Length: 214 > > v=0 > o=root 8523 8523 IN IP4 68.163.52.50 > s=session > c=IN IP4 68.163.52.50 > t=0 0 > m=audio 15640 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > (NAT) to 66.234.228.159:5060 > -- Called [EMAIL PROTECTED] > asterisk*CLI> > > Sip read: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 68.163.52.50:5060;branch=z9hG4bK600a4321;received=68.163.52.50;rport=50210 > From: "1000" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]>;tag=as1ecc3219 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Proxy-Authenticate: Digest realm="uasw001.voicepulse.com", nonce="5d626333" > Content-Length: 0 > > 11 headers, 0 lines > Transmitting: > ACK sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport > From: "1000" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]>;tag=as1ecc3219 > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 ACK > User-Agent: Asterisk PBX > Content-Length: 0 > > (NAT) to 66.234.228.159:5060 > We're at 68.163.52.50 port 15640 > Answering/Requesting with root capability 0x4 (ulaw) > Answering with non-codec capability 0x1 (telephone-event) > Reliably Transmitting: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport > From: "16109951010" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Proxy-Authorization: Digest username="s00******", > realm="uasw001.voicepulse.com", algorithm=MD5, > uri="sip:[EMAIL PROTECTED]", nonce="5d626333", > response="****HASH***", opaque="" > Date: Thu, 17 Feb 2005 22:10:02 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Type: application/sdp > Content-Length: 214 > > v=0 > o=root 8523 8524 IN IP4 68.163.52.50 > s=session > c=IN IP4 68.163.52.50 > t=0 0 > m=audio 15640 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > (NAT) to 66.234.228.159:5060 > asterisk*CLI> > > Sip read: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 68.163.52.50:5060;branch=z9hG4bK709030d1;received=68.163.52.50;rport=50210 > From: "16109951010" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]>;tag=as0630cede > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > 10 headers, 0 lines > asterisk*CLI> > > Sip read: > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP > 68.163.52.50:5060;branch=z9hG4bK709030d1;received=68.163.52.50;rport=50210 > From: "16109951010" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]>;tag=as0630cede > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Type: application/sdp > Content-Length: 373 > > v=0erisk*CLI> > o=root 24964 24964 IN IP4 66.234.228.159 > s=session > c=IN IP4 66.234.228.159 > t=0 0 > m=audio 10602 RTP/AVP 0 8 3 110 97 2 5 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:5 DVI4/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > 11 headers, 16 lines > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 3 > Found RTP audio format 110 > Found RTP audio format 97 > Found RTP audio format 2 > Found RTP audio format 5 > Found RTP audio format 101 > Peer audio RTP is at port 66.234.228.159:10602 > Found description format PCMU > Found description format PCMA > Found description format GSM > Found description format speex > Found description format iLBC > Found description format G726-32 > Found description format DVI4 > Found description format telephone-event > Capabilities: us - 0x4 (ulaw), peer - audio=0x63e > (gsm|ulaw|alaw|g726|adpcm|speex|ilbc)/video=0x0 (nothing), combined - > 0x4 (ulaw) > Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - > 0x1 (g723) > -- SIP/voicepulse-7990 is making progress passing it to SIP/1000-cd47 > We're at 192.168.1.102 port 11356 > Answering with preferred capability 0x4 (ulaw) > Answering with non-codec capability 0x1 (telephone-event) > Transmitting (no NAT): > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP 192.168.1.103:5061;branch=z9hG4bK-e7a8c127 > From: <sip:[EMAIL PROTECTED]>;tag=b0d057a1b98569abo1 > To: <sip:[EMAIL PROTECTED]>;tag=as7c26bda9 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Type: application/sdp > Content-Length: 216 > > v=0 > o=root 8523 8523 IN IP4 192.168.1.102 > s=session > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 11356 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > to 192.168.1.103:5061 > asterisk*CLI> > > 11 headers, 2 lines > Transmitting (no NAT): > SIP/2.0 200 OK > Via: SIP/2.0/UDP 66.234.228.159:5060;branch=z9hG4bK267fe14e > From: "voicepulse" <sip:[EMAIL PROTECTED]>;tag=as5cd2a689 > To: <sip:[EMAIL PROTECTED]>;tag=as47d60c4c > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Content-Length: 0 > > to 66.234.228.159:5060 > Destroying call '[EMAIL PROTECTED]' > asterisk*CLI> > > Sip read: > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP > 68.163.52.50:5060;branch=z9hG4bK709030d1;received=68.163.52.50;rport=50210 > From: "16109951010" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]>;tag=as0630cede > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > 10 headers, 0 lines > Transmitting: > ACK sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport > From: "1000" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]>;tag=as0630cede > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 103 ACK > User-Agent: Asterisk PBX > Content-Length: 0 > > (NAT) to 66.234.228.159:5060 > Feb 17 17:10:04 WARNING[8523]: chan_sip.c:6811 handle_response: > Forbidden - wrong password on authentication for INVITE to '"1000" > <sip:[EMAIL PROTECTED]>;tag=as74c56bff' > -- SIP/voicepulse-7990 is circuit-busy > Reliably Transmitting: > CANCEL sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport > From: "1000" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Proxy-Authorization: Digest username="s00******", > realm="uasw001.voicepulse.com", algorithm=MD5, > uri="sip:[EMAIL PROTECTED]", nonce="5d626333", > response="***HASH****", opaque="" > Content-Length: 0 > > (NAT) to 66.234.228.159:5060 > Scheduling destruction of call > '[EMAIL PROTECTED]' in 15000 ms > == Everyone is busy/congested at this time > -- Executing Hangup("SIP/1000-cd47", "") in new stack > == Spawn extension (macro-vp-dial, s, 2) exited non-zero on > 'SIP/1000-cd47' in macro 'vp-dial' > == Spawn extension (Home, 16109951010, 1) exited non-zero on 'SIP/1000-cd47' > Reliably Transmitting (no NAT): > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP 192.168.1.103:5061;branch=z9hG4bK-e7a8c127 > From: <sip:[EMAIL PROTECTED]>;tag=b0d057a1b98569abo1 > To: <sip:[EMAIL PROTECTED]>;tag=as7c26bda9 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > to 192.168.1.103:5061 > asterisk*CLI> > > Sip read: > SIP/2.0 481 Call Leg Does Not Exist > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1 > From: "1000" <sip:[EMAIL PROTECTED]>;tag=as74c56bff > To: <sip:[EMAIL PROTECTED]>;tag=as5baf064f > Call-ID: [EMAIL PROTECTED] > CSeq: 103 CANCEL > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Content-Length: 0 > > 10 headers, 0 lines > -- Got SIP response 481 "Call Leg Does Not Exist" back from 66.234.228.159 > Destroying call '[EMAIL PROTECTED]' > asterisk*CLI> > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users