vicidial.org

by **john_usc** » Fri Feb 04, 2011 12:09 am

my vicidial was working just fine and now suddenly I get call fail: temporary unavailable mesage. Can someone please help me out with this
I am using
goautodial
VERSION: 2.2.1-237
BUILD: 100510-2015
...................

here is the log

OPTIONS sip:cc119@72.215.73.192:5338;rinstance=9ab89864d35cea64;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK0e1bf741;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as5ce43f90
To: <sip:cc119@72.215.73.192:5338;rinstance=9ab89864d35cea64;cpd=on>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 1d8c260363b9a0a500c09557468e777f@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 04 Feb 2011 04:56:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Feb 3 23:56:18] VERBOSE[3022] logger.c: [Feb 3 23:56:18]
<--- SIP read from 72.215.73.192:5338 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK0e1bf741;rport=5060;received=67.73.36.181
Contact: <sip:72.215.73.192:5338>
To: <sip:cc119@72.215.73.192:5338;rinstance=9ab89864d35cea64;cpd=on>;tag=a273d40c
From: "asterisk"<sip:asterisk@192.168.1.2>;tag=as5ce43f90
Call-ID: 1d8c260363b9a0a500c09557468e777f@192.168.1.2
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
[Feb 3 23:56:18] VERBOSE[3022] logger.c: [Feb 3 23:56:18] --- (12 headers 0 lines) ---
[Feb 3 23:56:18] VERBOSE[3022] logger.c: [Feb 3 23:56:18] Really destroying SIP dialog '1d8c260363b9a0a500c09557468e777f@192.168.1.2' Method: OPTIONS
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23]
<--- SIP read from 72.215.73.192:5338 --->
INVITE sip:917201234567@67.73.36.181 SIP/2.0
Via: SIP/2.0/UDP 72.215.73.192:5338;branch=z9hG4bK-d8754z-b463b73bbb1ab54d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc119@72.215.73.192:5338>
To: "917201234567"<sip:917201234567@67.73.36.181>
From: "cc119"<sip:cc119@67.73.36.181>;tag=712a3f0b
Call-ID: YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 317

v=0
o=- 9 2 IN IP4 72.215.73.192
s=CounterPath X-Lite 3.0
c=IN IP4 72.215.73.192
t=0 0
m=audio 38144 RTP/AVP 107 0 8 101
a=alt:1 2 : d2hVH7de LwI5/V0L 5.162.52.103 38144
a=alt:2 1 : r/dKrOj1 hFaX0MAJ 192.168.1.3 38144
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] --- (12 headers 12 lines) ---
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Sending to 72.215.73.192 : 5338 (NAT)
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Using INVITE request as basis request - YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23]
<--- Reliably Transmitting (NAT) to 72.215.73.192:5338 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 72.215.73.192:5338;branch=z9hG4bK-d8754z-b463b73bbb1ab54d-1---d8754z-;received=72.215.73.192;rport=5338
From: "cc119"<sip:cc119@67.73.36.181>;tag=712a3f0b
To: "917201234567"<sip:917201234567@67.73.36.181>;tag=as3212935c
Call-ID: YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35e4d530"
Content-Length: 0

<------------>
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Scheduling destruction of SIP dialog 'YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.' in 32000 ms (Method: INVITE)
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Found user 'cc119'
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23]
<--- SIP read from 72.215.73.192:5338 --->
ACK sip:917201234567@67.73.36.181 SIP/2.0
Via: SIP/2.0/UDP 72.215.73.192:5338;branch=z9hG4bK-d8754z-b463b73bbb1ab54d-1---d8754z-;rport
To: "917201234567"<sip:917201234567@67.73.36.181>;tag=as3212935c
From: "cc119"<sip:cc119@67.73.36.181>;tag=712a3f0b
Call-ID: YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
CSeq: 1 ACK
Content-Length: 0

<------------->
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] --- (7 headers 0 lines) ---
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23]
<--- SIP read from 72.215.73.192:5338 --->
INVITE sip:917201234567@67.73.36.181 SIP/2.0
Via: SIP/2.0/UDP 72.215.73.192:5338;branch=z9hG4bK-d8754z-5e2bd340ff3b5a0e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc119@72.215.73.192:5338>
To: "917201234567"<sip:917201234567@67.73.36.181>
From: "cc119"<sip:cc119@67.73.36.181>;tag=712a3f0b
Call-ID: YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="cc119",realm="asterisk",nonce="35e4d530",uri="sip:917201234567@67.73.36.181",response="bb78270bc5f73c83c5c7e369d4caeda5",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 317

v=0
o=- 9 2 IN IP4 72.215.73.192
s=CounterPath X-Lite 3.0
c=IN IP4 72.215.73.192
t=0 0
m=audio 38144 RTP/AVP 107 0 8 101
a=alt:1 2 : d2hVH7de LwI5/V0L 5.162.52.103 38144
a=alt:2 1 : r/dKrOj1 hFaX0MAJ 192.168.1.3 38144
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] --- (13 headers 12 lines) ---
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Sending to 72.215.73.192 : 5338 (NAT)
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Using INVITE request as basis request - YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Found user 'cc119'
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Found RTP audio format 107
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Found RTP audio format 0
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Found RTP audio format 8
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Found RTP audio format 101
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Found unknown media description format BV32 for ID 107
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Found audio description format telephone-event for ID 101
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Peer audio RTP is at port 72.215.73.192:38144
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] Looking for 917201234567 in default (domain 67.73.36.181)
[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23] list_route: hop: <sip:cc119@72.215.73.192:5338>

[Feb 3 23:56:23] VERBOSE[3022] logger.c: [Feb 3 23:56:23]
<--- Transmitting (NAT) to 72.215.73.192:5338 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 72.215.73.192:5338;branch=z9hG4bK-d8754z-5e2bd340ff3b5a0e-1---d8754z-;received=72.215.73.192;rport=5338
From: "cc119"<sip:cc119@67.73.36.181>;tag=712a3f0b
To: "917201234567"<sip:917201234567@67.73.36.181>
Call-ID: YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917201234567@192.168.1.2>
Content-Length: 0

<------------>
[Feb 3 23:56:23] VERBOSE[20828] logger.c: [Feb 3 23:56:23] -- Executing [917201234567@default:1] AGI("SIP/cc119-00000010", "agi://127.0.0.1:4577/call_log") in new stack
[Feb 3 23:56:23] VERBOSE[20828] logger.c: [Feb 3 23:56:23] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb 3 23:56:23] VERBOSE[20828] logger.c: [Feb 3 23:56:23] -- Executing [917201234567@default:2] Dial("SIP/cc119-00000010", "SIP/goautodial/17291234657||tTor") in new stack
[Feb 3 23:56:27] VERBOSE[3022] logger.c: [Feb 3 23:56:27]
<--- SIP read from 72.215.73.192:5338 --->

<------------->
[Feb 3 23:56:57] VERBOSE[3022] logger.c: [Feb 3 23:56:57]
<--- SIP read from 72.215.73.192:5338 --->

<------------->
[Feb 3 23:57:01] VERBOSE[20926] logger.c: [Feb 3 23:57:01] == Parsing '/etc/asterisk/manager.conf': [Feb 3 23:57:01] VERBOSE[20926] logger.c: [Feb 3 23:57:01] Found
[Feb 3 23:57:01] VERBOSE[20926] logger.c: [Feb 3 23:57:01] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 3 23:57:02] VERBOSE[20926] logger.c: [Feb 3 23:57:02] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 3 23:57:02] VERBOSE[20937] logger.c: [Feb 3 23:57:02] == Parsing '/etc/asterisk/manager.conf': [Feb 3 23:57:02] VERBOSE[20937] logger.c: [Feb 3 23:57:02] Found
[Feb 3 23:57:02] VERBOSE[20937] logger.c: [Feb 3 23:57:02] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 3 23:57:07] VERBOSE[20949] logger.c: [Feb 3 23:57:07] == Parsing '/etc/asterisk/manager.conf': [Feb 3 23:57:07] VERBOSE[20949] logger.c: [Feb 3 23:57:07] Found
nd
[Feb 3 23:57:02] VERBOSE[20937] logger.c: [Feb 3 23:57:02] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 3 23:57:07] VERBOSE[20949] logger.c: [Feb 3 23:57:07] == Parsing '/etc/asterisk/manager.conf': [Feb 3 23:57:07] VERBOSE[20949] logger.c: [Feb 3 23:57:07] Found
[Feb 3 23:57:07] VERBOSE[20949] logger.c: [Feb 3 23:57:07] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 3 23:57:07] VERBOSE[20949] logger.c: [Feb 3 23:57:07] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 3 23:57:07] ERROR[20937] utils.c: write() returned error: Broken pipe
[Feb 3 23:57:07] VERBOSE[20937] logger.c: [Feb 3 23:57:07] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 3 23:57:18] VERBOSE[3022] logger.c: [Feb 3 23:57:18] Reliably Transmitting (NAT) to 72.215.73.192:5338:
OPTIONS sip:cc119@72.215.73.192:5338;rinstance=9ab89864d35cea64;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK18e6a62d;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as450401b3
To: <sip:cc119@72.215.73.192:5338;rinstance=9ab89864d35cea64;cpd=on>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 093cb2ff5466799a322ff7ff00a6c334@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 04 Feb 2011 04:57:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Feb 3 23:57:18] VERBOSE[3022] logger.c: [Feb 3 23:57:18]
<--- SIP read from 72.215.73.192:5338 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK18e6a62d;rport=5060;received=67.73.36.181
Contact: <sip:72.215.73.192:5338>
To: <sip:cc119@72.215.73.192:5338;rinstance=9ab89864d35cea64;cpd=on>;tag=8e555738
From: "asterisk"<sip:asterisk@192.168.1.2>;tag=as450401b3
Call-ID: 093cb2ff5466799a322ff7ff00a6c334@192.168.1.2
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
<------------->
[Feb 3 23:57:18] VERBOSE[3022] logger.c: [Feb 3 23:57:18] --- (12 headers 0 lines) ---
[Feb 3 23:57:18] VERBOSE[3022] logger.c: [Feb 3 23:57:18] Really destroying SIP dialog '093cb2ff5466799a322ff7ff00a6c334@192.168.1.2' Method: OPTIONS
[Feb 3 23:57:23] WARNING[20828] chan_sip.c: No such host: goautodial
[Feb 3 23:57:23] WARNING[20828] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Feb 3 23:57:23] VERBOSE[20828] logger.c: [Feb 3 23:57:23] == Everyone is busy/congested at this time (1:0/0/1)
[Feb 3 23:57:23] VERBOSE[20828] logger.c: [Feb 3 23:57:23] -- Executing [917201234567@default:3] Hangup("SIP/cc119-00000010", "") in new stack
[Feb 3 23:57:23] VERBOSE[20828] logger.c: [Feb 3 23:57:23] == Spawn extension (default, 917201234567, 3) exited non-zero on 'SIP/cc119-00000010'
[Feb 3 23:57:23] VERBOSE[20828] logger.c: [Feb 3 23:57:23] -- Executing [h@default:1] DeadAGI("SIP/cc119-00000010", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Feb 3 23:57:23] VERBOSE[3022] logger.c: [Feb 3 23:57:23]
<--- Transmitting (NAT) to 72.215.73.192:5338 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 72.215.73.192:5338;branch=z9hG4bK-d8754z-5e2bd340ff3b5a0e-1---d8754z-;received=72.215.73.192;rport=5338
From: "cc119"<sip:cc119@67.73.36.181>;tag=712a3f0b
To: "917201234567"<sip:917201234567@67.73.36.181>;tag=as504fb824
Call-ID: YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917201234567@192.168.1.2>
Content-Length: 0

<------------>
[Feb 3 23:57:23] VERBOSE[20828] logger.c: [Feb 3 23:57:23] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Feb 3 23:57:23] VERBOSE[20828] logger.c: [Feb 3 23:57:23] Scheduling destruction of SIP dialog 'YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.' in 32000 ms (Method: INVITE)
[Feb 3 23:57:23] VERBOSE[20828] logger.c: [Feb 3 23:57:23]
<--- Reliably Transmitting (NAT) to 72.215.73.192:5338 --->
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 72.215.73.192:5338;branch=z9hG4bK-d8754z-5e2bd340ff3b5a0e-1---d8754z-;received=72.215.73.192;rport=5338
From: "cc119"<sip:cc119@67.73.36.181>;tag=712a3f0b
To: "917201234567"<sip:917201234567@67.73.36.181>;tag=as504fb824
Call-ID: YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Feb 3 23:57:23] VERBOSE[3022] logger.c: [Feb 3 23:57:23]
<--- SIP read from 72.215.73.192:5338 --->
ACK sip:917201234567@67.73.36.181 SIP/2.0
Via: SIP/2.0/UDP 72.215.73.192:5338;branch=z9hG4bK-d8754z-5e2bd340ff3b5a0e-1---d8754z-;rport
To: "917201234567"<sip:917201234567@67.73.36.181>;tag=as504fb824
From: "cc119"<sip:cc119@67.73.36.181>;tag=712a3f0b
Call-ID: YjRiYTE1YjAwYzIxNTZkNGUzNzQwNWE2NjUyZTI4YzU.
CSeq: 2 ACK
Content-Length: 0

<------------->
[Feb 3 23:57:23] VERBOSE[3022] logger.c: [Feb 3 23:57:23] --- (7 headers 0 lines) ---
[Feb 3 23:57:27] VERBOSE[3022] logger.c: [Feb 3 23:57:27]
<--- SIP read from 72.215.73.192:5338 --->

<------------->
[Feb 3 23:57:40] VERBOSE[20702] logger.c: [Feb 3 23:57:40] -- Remote UNIX connection disconnected

by **okli** » Fri Feb 04, 2011 1:48 am

[Feb 3 23:56:23] VERBOSE[20828] logger.c: [Feb 3 23:56:23] -- Executing [917201234567@default:2] Dial("SIP/cc119-00000010", "SIP/goautodial/17291234657||tTor") in new stack

....

[Feb 3 23:57:23] WARNING[20828] chan_sip.c: No such host: goautodial
[Feb 3 23:57:23] WARNING[20828] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Feb 3 23:57:23] VERBOSE[20828] logger.c: [Feb 3 23:57:23] == Everyone is busy/congested at this time (1:0/0/1)

Check settings.

Ans please do use code or quote tags, especially when posting such lengthy logs.
Also post verbose logs only if have to. This would greatly increase your chances someone to read it.

by **john_usc** » Fri Feb 04, 2011 9:20 am

so I am not sure whats happening here. How can I fix it ? any idea?

by **okli** » Fri Feb 04, 2011 12:54 pm

Just check your carrier settings. Which one is it? How did you set it up?

by **john_usc** » Fri Feb 04, 2011 2:42 pm

it is vitelity. I have ip authentication. I talked to those guys and they said it was something in my dialer...may be extension.conf or something.

by **gardo** » Fri Feb 04, 2011 2:51 pm

What's your carrier entry for Vitelity? You might still be using our default carrier entry.

by **john_usc** » Fri Feb 04, 2011 4:31 pm

Gardo

I kept the trunk name same as default thats why its showing like that. But the enteries I used are for vitelity and not for goautodial.
I am pulling my hair as all my guys are sitting and waiting.... we are down

by **okli** » Fri Feb 04, 2011 4:44 pm

Unless you provide more details how and where you set your provider, what the entries are, as you were asked already twice, don't know how to help you any further...

by **john_usc** » Fri Feb 04, 2011 5:29 pm

here are my settings

CARRIERNAME
EXS

ACCOUNT ENTRY
[EXS]
disallow=all
allow=ulaw
type=friend
trustrpid=yes
sendrpid=yes
host=64.2.142.93
dtmfmode=auto
canreinvite=no
insecure=very

GLOBAL STRING
TRUNK = SIP/EXS

DIALPLAN Entry

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTor)
exten => _91NXXNXXXXXX,3,Hangup

by **john_usc** » Fri Feb 04, 2011 5:39 pm

ok I just reinstalled goautodial. The latest version
VERSION 2.2.1-237
BUILD: 100510-2015

and I used the default settings of goautodial now and opened an account of 0.5 that they give you initially to test the call. I tried to dial the number and its same temporary unavialable message!!!

by **okli** » Fri Feb 04, 2011 6:09 pm

Can you post now some asterisk output, no verbose please.

by **john_usc** » Fri Feb 04, 2011 6:45 pm

here is the cli

[Feb 4 18:43:35] NOTICE[3572]: chan_sip.c:13408 handle_response_peerpoke: Peer 'cc100' is now Reachable. (13ms / 2000ms)
[Feb 4 18:43:53] -- Executing [917293834792@default:1] AGI("SIP/cc100-00000002", "agi://127.0.0.1:4577/call_log") in new stack
[Feb 4 18:43:53] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb 4 18:43:53] -- Executing [917293834792@default:2] Dial("SIP/cc100-00000002", "SIP/goautodial/917293834792||tTor") in new stack
[Feb 4 18:43:53] WARNING[27658]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Feb 4 18:43:53] == Everyone is busy/congested at this time (1:0/0/1)
[Feb 4 18:43:53] -- Executing [917293834792@default:3] Hangup("SIP/cc100-00000002", "") in new stack
[Feb 4 18:43:53] == Spawn extension (default, 917203334762, 3) exited non-zero on 'SIP/cc100-00000002'
[Feb 4 18:43:53] -- Executing [h@default:1] DeadAGI("SIP/cc100-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Feb 4 18:43:53] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Feb 4 18:44:01] == Parsing '/etc/asterisk/manager.conf': [Feb 4 18:44:01] Found
[Feb 4 18:44:01] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 4 18:44:01] == Parsing '/etc/asterisk/manager.conf': [Feb 4 18:44:01] Found
[Feb 4 18:44:01] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 4 18:44:01] == Manager 'sendcron' logged off from 127.0.0.1

[/quote]

by **okli** » Fri Feb 04, 2011 7:06 pm

It's still dialling through goautodial entry.

Executing [917293834792@default:2] Dial("SIP/cc100-00000002", "SIP/goautodial/917293834792||tTor") in new stack

Check in extensions.conf and admin interface-->carriers for goautodial entries and remove them.

by **john_usc** » Fri Feb 04, 2011 7:26 pm

ahhh..found it...

we need to have context=default

why is it not included in the default download I wonder?
anyhow thanks you guys..see you guys soon:D

[sipTrunks]
disallow=all
allow=ulaw
type=friend
trustrpid=yes
sendrpid=yes
host=64.2.142.93
dtmfmode=auto
canreinvite=no
context=default

vicidial.org

call temproray unavailable

call temproray unavailable

Who is online