[Asterisk-Users] Re: Voicepulse Open Access & Asterisk Problems
Brian Dingman
bdingman at gmail.com
Mon Mar 14 17:24:33 MST 2005
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 <bdingman at gmail.com> 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******:********@access1.voicepulse.com
>
> 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/16109951010 at voicepulse") in new stack
> -- Called 16109951010 at voicepulse
> -- 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:1000 at 68.163.52.50>;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/16109951010 at voicepulse") 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:16109951010 at access1.voicepulse.com SIP/2.0
> Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport
> From: "1000" <sip:1000 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>
> Contact: <sip:1000 at 68.163.52.50>
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> 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 16109951010 at voicepulse
> 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:1000 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>;tag=as1ecc3219
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> CSeq: 102 INVITE
> User-Agent: VoicePulse SW
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:16109951010 at 66.234.228.159>
> Proxy-Authenticate: Digest realm="uasw001.voicepulse.com", nonce="5d626333"
> Content-Length: 0
>
> 11 headers, 0 lines
> Transmitting:
> ACK sip:16109951010 at access1.voicepulse.com SIP/2.0
> Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport
> From: "1000" <sip:1000 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>;tag=as1ecc3219
> Contact: <sip:1000 at 68.163.52.50>
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> 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:16109951010 at access1.voicepulse.com SIP/2.0
> Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport
> From: "16109951010" <sip:16109951010 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>
> Contact: <sip:16109951010 at 68.163.52.50>
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Proxy-Authorization: Digest username="s00******",
> realm="uasw001.voicepulse.com", algorithm=MD5,
> uri="sip:16109951010 at 66.234.228.159", 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:16109951010 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>;tag=as0630cede
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> CSeq: 103 INVITE
> User-Agent: VoicePulse SW
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:16109951010 at 66.234.228.159>
> 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:16109951010 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>;tag=as0630cede
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> CSeq: 103 INVITE
> User-Agent: VoicePulse SW
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:16109951010 at 66.234.228.159>
> 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:1000 at 192.168.1.102>;tag=b0d057a1b98569abo1
> To: <sip:16109951010 at 192.168.1.102>;tag=as7c26bda9
> Call-ID: 8ddc2f59-c7e8b553 at 192.168.1.103
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:16109951010 at 192.168.1.102>
> 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:voicepulse at 66.234.228.159>;tag=as5cd2a689
> To: <sip:s at 68.163.52.50>;tag=as47d60c4c
> Call-ID: 21756c3462a4711e132bd1d1668184ab at 66.234.228.159
> 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 '21756c3462a4711e132bd1d1668184ab at 66.234.228.159'
> 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:16109951010 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>;tag=as0630cede
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> CSeq: 103 INVITE
> User-Agent: VoicePulse SW
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:16109951010 at 66.234.228.159>
> Content-Length: 0
>
> 10 headers, 0 lines
> Transmitting:
> ACK sip:16109951010 at access1.voicepulse.com SIP/2.0
> Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport
> From: "1000" <sip:1000 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>;tag=as0630cede
> Contact: <sip:16109951010 at 68.163.52.50>
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> 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:1000 at 68.163.52.50>;tag=as74c56bff'
> -- SIP/voicepulse-7990 is circuit-busy
> Reliably Transmitting:
> CANCEL sip:16109951010 at access1.voicepulse.com SIP/2.0
> Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport
> From: "1000" <sip:1000 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>
> Contact: <sip:16109951010 at 68.163.52.50>
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> CSeq: 103 CANCEL
> User-Agent: Asterisk PBX
> Proxy-Authorization: Digest username="s00******",
> realm="uasw001.voicepulse.com", algorithm=MD5,
> uri="sip:16109951010 at 66.234.228.159", nonce="5d626333",
> response="***HASH****", opaque=""
> Content-Length: 0
>
> (NAT) to 66.234.228.159:5060
> Scheduling destruction of call
> '7575529303e8335959625cd640e68ca2 at 68.163.52.50' 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:1000 at 192.168.1.102>;tag=b0d057a1b98569abo1
> To: <sip:16109951010 at 192.168.1.102>;tag=as7c26bda9
> Call-ID: 8ddc2f59-c7e8b553 at 192.168.1.103
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:16109951010 at 192.168.1.102>
> 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:1000 at 68.163.52.50>;tag=as74c56bff
> To: <sip:16109951010 at access1.voicepulse.com>;tag=as5baf064f
> Call-ID: 7575529303e8335959625cd640e68ca2 at 68.163.52.50
> 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 '7575529303e8335959625cd640e68ca2 at 68.163.52.50'
> asterisk*CLI>
>
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