Brian Dingman
2005-Feb-17 17:03 UTC
[Asterisk-Users] Voicepulse Open Access & Asterisk Problems
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@voicepulse") in new stack
-- Called 16109951010@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@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@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@access1.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport
From: "1000" <sip:1000@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>
Contact: <sip:1000@68.163.52.50>
Call-ID: 7575529303e8335959625cd640e68ca2@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@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@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>;tag=as1ecc3219
Call-ID: 7575529303e8335959625cd640e68ca2@68.163.52.50
CSeq: 102 INVITE
User-Agent: VoicePulse SW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:16109951010@66.234.228.159>
Proxy-Authenticate: Digest realm="uasw001.voicepulse.com",
nonce="5d626333"
Content-Length: 0
11 headers, 0 lines
Transmitting:
ACK sip:16109951010@access1.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport
From: "1000" <sip:1000@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>;tag=as1ecc3219
Contact: <sip:1000@68.163.52.50>
Call-ID: 7575529303e8335959625cd640e68ca2@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@access1.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport
From: "16109951010"
<sip:16109951010@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>
Contact: <sip:16109951010@68.163.52.50>
Call-ID: 7575529303e8335959625cd640e68ca2@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@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@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>;tag=as0630cede
Call-ID: 7575529303e8335959625cd640e68ca2@68.163.52.50
CSeq: 103 INVITE
User-Agent: VoicePulse SW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:16109951010@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@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>;tag=as0630cede
Call-ID: 7575529303e8335959625cd640e68ca2@68.163.52.50
CSeq: 103 INVITE
User-Agent: VoicePulse SW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:16109951010@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@192.168.1.102>;tag=b0d057a1b98569abo1
To: <sip:16109951010@192.168.1.102>;tag=as7c26bda9
Call-ID: 8ddc2f59-c7e8b553@192.168.1.103
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:16109951010@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@66.234.228.159>;tag=as5cd2a689
To: <sip:s@68.163.52.50>;tag=as47d60c4c
Call-ID: 21756c3462a4711e132bd1d1668184ab@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@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@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>;tag=as0630cede
Call-ID: 7575529303e8335959625cd640e68ca2@68.163.52.50
CSeq: 103 INVITE
User-Agent: VoicePulse SW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:16109951010@66.234.228.159>
Content-Length: 0
10 headers, 0 lines
Transmitting:
ACK sip:16109951010@access1.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport
From: "1000" <sip:1000@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>;tag=as0630cede
Contact: <sip:16109951010@68.163.52.50>
Call-ID: 7575529303e8335959625cd640e68ca2@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@68.163.52.50>;tag=as74c56bff'
-- SIP/voicepulse-7990 is circuit-busy
Reliably Transmitting:
CANCEL sip:16109951010@access1.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport
From: "1000" <sip:1000@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>
Contact: <sip:16109951010@68.163.52.50>
Call-ID: 7575529303e8335959625cd640e68ca2@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@66.234.228.159", nonce="5d626333",
response="***HASH****", opaque=""
Content-Length: 0
(NAT) to 66.234.228.159:5060
Scheduling destruction of call
'7575529303e8335959625cd640e68ca2@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@192.168.1.102>;tag=b0d057a1b98569abo1
To: <sip:16109951010@192.168.1.102>;tag=as7c26bda9
Call-ID: 8ddc2f59-c7e8b553@192.168.1.103
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:16109951010@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@68.163.52.50>;tag=as74c56bff
To: <sip:16109951010@access1.voicepulse.com>;tag=as5baf064f
Call-ID: 7575529303e8335959625cd640e68ca2@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@68.163.52.50'
asterisk*CLI>
Brian Dingman
2005-Mar-14 17:24 UTC
[Asterisk-Users] Re: Voicepulse Open Access & Asterisk Problems
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@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@voicepulse") in new stack > -- Called 16109951010@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@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@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@access1.voicepulse.com SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport > From: "1000" <sip:1000@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com> > Contact: <sip:1000@68.163.52.50> > Call-ID: 7575529303e8335959625cd640e68ca2@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@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@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com>;tag=as1ecc3219 > Call-ID: 7575529303e8335959625cd640e68ca2@68.163.52.50 > CSeq: 102 INVITE > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:16109951010@66.234.228.159> > Proxy-Authenticate: Digest realm="uasw001.voicepulse.com", nonce="5d626333" > Content-Length: 0 > > 11 headers, 0 lines > Transmitting: > ACK sip:16109951010@access1.voicepulse.com SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport > From: "1000" <sip:1000@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com>;tag=as1ecc3219 > Contact: <sip:1000@68.163.52.50> > Call-ID: 7575529303e8335959625cd640e68ca2@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@access1.voicepulse.com SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport > From: "16109951010" <sip:16109951010@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com> > Contact: <sip:16109951010@68.163.52.50> > Call-ID: 7575529303e8335959625cd640e68ca2@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@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@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com>;tag=as0630cede > Call-ID: 7575529303e8335959625cd640e68ca2@68.163.52.50 > CSeq: 103 INVITE > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:16109951010@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@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com>;tag=as0630cede > Call-ID: 7575529303e8335959625cd640e68ca2@68.163.52.50 > CSeq: 103 INVITE > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:16109951010@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@192.168.1.102>;tag=b0d057a1b98569abo1 > To: <sip:16109951010@192.168.1.102>;tag=as7c26bda9 > Call-ID: 8ddc2f59-c7e8b553@192.168.1.103 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:16109951010@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@66.234.228.159>;tag=as5cd2a689 > To: <sip:s@68.163.52.50>;tag=as47d60c4c > Call-ID: 21756c3462a4711e132bd1d1668184ab@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@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@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com>;tag=as0630cede > Call-ID: 7575529303e8335959625cd640e68ca2@68.163.52.50 > CSeq: 103 INVITE > User-Agent: VoicePulse SW > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:16109951010@66.234.228.159> > Content-Length: 0 > > 10 headers, 0 lines > Transmitting: > ACK sip:16109951010@access1.voicepulse.com SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport > From: "1000" <sip:1000@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com>;tag=as0630cede > Contact: <sip:16109951010@68.163.52.50> > Call-ID: 7575529303e8335959625cd640e68ca2@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@68.163.52.50>;tag=as74c56bff' > -- SIP/voicepulse-7990 is circuit-busy > Reliably Transmitting: > CANCEL sip:16109951010@access1.voicepulse.com SIP/2.0 > Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport > From: "1000" <sip:1000@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com> > Contact: <sip:16109951010@68.163.52.50> > Call-ID: 7575529303e8335959625cd640e68ca2@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@66.234.228.159", nonce="5d626333", > response="***HASH****", opaque="" > Content-Length: 0 > > (NAT) to 66.234.228.159:5060 > Scheduling destruction of call > '7575529303e8335959625cd640e68ca2@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@192.168.1.102>;tag=b0d057a1b98569abo1 > To: <sip:16109951010@192.168.1.102>;tag=as7c26bda9 > Call-ID: 8ddc2f59-c7e8b553@192.168.1.103 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:16109951010@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@68.163.52.50>;tag=as74c56bff > To: <sip:16109951010@access1.voicepulse.com>;tag=as5baf064f > Call-ID: 7575529303e8335959625cd640e68ca2@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@68.163.52.50' > asterisk*CLI> >