[Asterisk-Users] Voicepulse Open Access & Asterisk Problems

Brian Dingman bdingman at gmail.com
Thu Feb 17 17:03:39 MST 2005 

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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