Hi all,
Goautodial 2.1
version 2.4-309a
build:110430-1642
asterisk 1.4.39.1
from iso.
no extra hardware.
no addons.
using eybeam as a softphone.
one agent loging for the test.
p3 computer 930 Mhz 256 RAM just to test goautodial
Can't make calls using goautodial.installed it successfully without errors.
Created compagnes added lists and added users. and also carrier.
Can't make calls using predictive dialing.even though there are calls on the server.
not even when using manuel dial methode.
but i can successfully make call using my softphone either using my server configuration or my provider's.
i have made no modifacations to the server apart from changing its ip .but i did run the update server command.
also to mention that when connecting to the server in both manuel and predictive i get the message :you are the only person ...
when using the manuel dial methode i get the following message:
DIAL ALERT.
CALL REJECTED BUSY
CAUSE:21 -CALL REJECTED
Below is the cli when using manuel dialing
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [58600051@default:1] MeetMe("Local/58600051@default-9d94,2", "8600051|Fmq") in new stack
-- Executing [8309@default:1] Answer("Local/58600051@default-9d94,1", "") in new stack
-- Executing [8309@default:2] Monitor("Local/58600051@default-9d94,1", "wav|20110703-164625_0386572380") in new stack
-- Executing [8309@default:3] Wait("Local/58600051@default-9d94,1", "3600") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
**********************************************
Below is the cli when using predictive dialing.
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [90386583210@default:1] AGI("Local/90386583210@default-3bcb,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [90386583210@default:2] Dial("Local/90386583210@default-3bcb,2", "SIP/siptrunk/90386583210||To") in new stack
-- Called siptrunk/90386583210
-- Got SIP response 603 "Declined" back from 46.182.3.50
-- SIP/siptrunk-00000173 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [90386583210@default:3] Hangup("Local/90386583210@default-3bcb,2", "") in new stack
== Spawn extension (default, 90386583210, 3) exited non-zero on 'Local/90386583210@default-3bcb,2'
-- Executing [h@default:1] DeadAGI("Local/90386583210@default-3bcb,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [90386615913@default:1] AGI("Local/90386615913@default-a370,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [90386615913@default:2] Dial("Local/90386615913@default-a370,2", "SIP/siptrunk/90386615913||To") in new stack
-- Called siptrunk/90386615913
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
-- Got SIP response 603 "Declined" back from 46.182.3.50
-- SIP/siptrunk-00000174 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [90386615913@default:3] Hangup("Local/90386615913@default-a370,2", "") in new stack
== Spawn extension (default, 90386615913, 3) exited non-zero on 'Local/90386615913@default-a370,2'
-- Executing [h@default:1] DeadAGI("Local/90386615913@default-a370,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [90386574435@default:1] AGI("Local/90386574435@default-3e1a,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [90386574435@default:2] Dial("Local/90386574435@default-3e1a,2", "SIP/siptrunk/90386574435||To") in new stack
-- Called siptrunk/90386574435
-- Got SIP response 603 "Declined" back from 46.182.3.50
-- SIP/siptrunk-00000175 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [90386574435@default:3] Hangup("Local/90386574435@default-3e1a,2", "") in new stack
== Spawn extension (default, 90386574435, 3) exited non-zero on 'Local/90386574435@default-3e1a,2'
-- Executing [h@default:1] DeadAGI("Local/90386574435@default-3e1a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [90386285041@default:1] AGI("Local/90386285041@default-6c3e,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [90386285041@default:2] Dial("Local/90386285041@default-6c3e,2", "SIP/siptrunk/90386285041||To") in new stack
-- Called siptrunk/90386285041
-- Got SIP response 603 "Declined" back from 46.182.3.50
-- SIP/siptrunk-00000176 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [90386285041@default:3] Hangup("Local/90386285041@default-6c3e,2", "") in new stack
== Spawn extension (default, 90386285041, 3) exited non-zero on 'Local/90386285041@default-6c3e,2'
-- Executing [h@default:1] DeadAGI("Local/90386285041@default-6c3e,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [90386784004@default:1] AGI("Local/90386784004@default-9140,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [90386784004@default:2] Dial("Local/90386784004@default-9140,2", "SIP/siptrunk/90386784004||To") in new stack
-- Called siptrunk/90386784004
-- Got SIP response 603 "Declined" back from 46.182.3.50
-- SIP/siptrunk-00000177 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [90386784004@default:3] Hangup("Local/90386784004@default-9140,2", "") in new stack
== Spawn extension (default, 90386784004, 3) exited non-zero on 'Local/90386784004@default-9140,2'
-- Executing [h@default:1] DeadAGI("Local/90386784004@default-9140,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [90386377850@default:1] AGI("Local/90386377850@default-b3a2,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [90386377850@default:2] Dial("Local/90386377850@default-b3a2,2", "SIP/siptrunk/90386377850||To") in new stack
-- Called siptrunk/90386377850
-- AGI Script agi://127.0.0.1:4577/call_log-
***********************
Max Calls per Second:20
*************************
phone numbers are put like this in the lists
0142924292(calling from MOROCCO to FRANCE)
dialplan
exten => _01.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _01.,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _01.,3,Hangup
exten => _02.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _02.,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _02.,3,Hangup
exten => _03.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _03.,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _03.,3,Hangup
exten => _04.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _04.,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _04.,3,Hangup
exten => _05.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _05.,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _05.,3,Hangup
exten => _08.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _08.,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _08.,3,Hangup
exten => _09.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _09.,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _09.,3,Hangup
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(${TRUNK}/${EXTEN},,To)
exten => _X.,3,Hangup
************************
we have the same configuration in a call center and it works great.(with both dialplan and carrier.
*****************************
Globals String :TRUNK = SIP/siptrunk
************
carrier.
[siptrunk]
host=ip
type=friend
context=from-trunk
username=myusername
secret=mysecret
fromuser=myusername
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=g729
insecure=very
nat=yes
**************************
i can add that my processor is overloaded
top command:
top - 17:27:11 up 3:11, 2 users, load average: 1.31, 1.13, 1.14
Tasks: 123 total, 1 running, 122 sleeping, 0 stopped, 0 zombie
Cpu(s): 22.0%us, 14.7%sy, 0.0%ni, 63.3%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st
Mem: 254004k total, 233772k used, 20232k free, 9992k buffers
Swap: 524280k total, 92k used, 524188k free, 78480k cached
******************
hope i gave lot of information.
thank you
Can't make calls using Goautodial
Moderators: enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, s0lid
8 posts
• Page 1 of 1
Can't make calls using Goautodial
by sofcall » Sun Jul 03, 2011 10:31 am
- sofcall
- Posts: 110
- Joined: Wed Feb 02, 2011 4:49 am
- Location: Morocco
by williamconley » Sun Jul 03, 2011 2:18 pm
this is your carrier rejecting the call. if you use sip debug, you may get a more detailed message, but you may not.Got SIP response 603 "Declined" back from 46.182.3.50
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
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http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
- williamconley
- Posts: 20258
- Joined: Wed Oct 31, 2007 4:17 pm
- Location: Davenport, FL (By Disney!)
by sofcall » Sun Jul 03, 2011 4:15 pm
sip debug
go*CLI> sip debug
SIP Debugging re-enabled
[Jul 3 23:13:19] Reliably Transmitting (NAT) to 46.182.3.50:5060:
OPTIONS sip:46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK5f0a3d9e;rport
From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as0486e673
To: <sip:46.182.3.50;cpd=on>
Contact: <sip:asterisk@192.168.1.3>
Call-ID: 2f222df4123de1b208ddf4000c4920ee@192.168.1.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 03 Jul 2011 21:13:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
---
[Jul 3 23:13:19]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK5f0a3d9e;received=192.168.1.3;rport=49586
From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as0486e673
To: <sip:46.182.3.50;cpd=on>;tag=as6e6b762f
Call-ID: 2f222df4123de1b208ddf4000c4920ee@192.168.1.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:46.182.3.50:5060>
Accept: application/sdp
Content-Length: 0
<------------->
[Jul 3 23:13:19] --- (12 headers 0 lines) ---
[Jul 3 23:13:19] Really destroying SIP dialog '2f222df4123de1b208ddf4000c4920ee@192.168.1.3' Method: OPTIONS
[Jul 3 23:13:25] Reliably Transmitting (NAT) to 192.168.1.74:37264:
OPTIONS sip:cc102@192.168.1.74:37264;rinstance=1153d06a3c93e65b;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK052c634a;rport
From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as1e378016
To: <sip:cc102@192.168.1.74:37264;rinstance=1153d06a3c93e65b;cpd=on>
Contact: <sip:asterisk@192.168.1.3>
Call-ID: 294d42a9011a10815bc8c327511afe67@192.168.1.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 03 Jul 2011 21:13:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
---
[Jul 3 23:13:25]
<--- SIP read from 192.168.1.74:37264 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK052c634a;rport=5060
Contact: <sip:192.168.1.74:37264>
To: <sip:cc102@192.168.1.74:37264;rinstance=1153d06a3c93e65b;cpd=on>;tag=c2301474
From: "asterisk"<sip:asterisk@192.168.1.3>;tag=as1e378016
Call-ID: 294d42a9011a10815bc8c327511afe67@192.168.1.3
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0
<------------->
[Jul 3 23:13:25] --- (12 headers 0 lines) ---
[Jul 3 23:13:25]
<--- SIP read from 192.168.1.74:37264 --->
<------------->
[Jul 3 23:13:25] Really destroying SIP dialog '294d42a9011a10815bc8c327511afe67@192.168.1.3' Method: OPTIONS
go*CLI>
go*CLI> sip debug
SIP Debugging re-enabled
[Jul 3 23:13:19] Reliably Transmitting (NAT) to 46.182.3.50:5060:
OPTIONS sip:46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK5f0a3d9e;rport
From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as0486e673
To: <sip:46.182.3.50;cpd=on>
Contact: <sip:asterisk@192.168.1.3>
Call-ID: 2f222df4123de1b208ddf4000c4920ee@192.168.1.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 03 Jul 2011 21:13:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
---
[Jul 3 23:13:19]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK5f0a3d9e;received=192.168.1.3;rport=49586
From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as0486e673
To: <sip:46.182.3.50;cpd=on>;tag=as6e6b762f
Call-ID: 2f222df4123de1b208ddf4000c4920ee@192.168.1.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:46.182.3.50:5060>
Accept: application/sdp
Content-Length: 0
<------------->
[Jul 3 23:13:19] --- (12 headers 0 lines) ---
[Jul 3 23:13:19] Really destroying SIP dialog '2f222df4123de1b208ddf4000c4920ee@192.168.1.3' Method: OPTIONS
[Jul 3 23:13:25] Reliably Transmitting (NAT) to 192.168.1.74:37264:
OPTIONS sip:cc102@192.168.1.74:37264;rinstance=1153d06a3c93e65b;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK052c634a;rport
From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as1e378016
To: <sip:cc102@192.168.1.74:37264;rinstance=1153d06a3c93e65b;cpd=on>
Contact: <sip:asterisk@192.168.1.3>
Call-ID: 294d42a9011a10815bc8c327511afe67@192.168.1.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 03 Jul 2011 21:13:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
---
[Jul 3 23:13:25]
<--- SIP read from 192.168.1.74:37264 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK052c634a;rport=5060
Contact: <sip:192.168.1.74:37264>
To: <sip:cc102@192.168.1.74:37264;rinstance=1153d06a3c93e65b;cpd=on>;tag=c2301474
From: "asterisk"<sip:asterisk@192.168.1.3>;tag=as1e378016
Call-ID: 294d42a9011a10815bc8c327511afe67@192.168.1.3
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0
<------------->
[Jul 3 23:13:25] --- (12 headers 0 lines) ---
[Jul 3 23:13:25]
<--- SIP read from 192.168.1.74:37264 --->
<------------->
[Jul 3 23:13:25] Really destroying SIP dialog '294d42a9011a10815bc8c327511afe67@192.168.1.3' Method: OPTIONS
go*CLI>
- sofcall
- Posts: 110
- Joined: Wed Feb 02, 2011 4:49 am
- Location: Morocco
by williamconley » Sun Jul 03, 2011 5:16 pm
Interesting, but none of that was one of the rejected calls.
If you check the cli, the "Declined" message will be right AFTER the sip debug that caused it. So post the sip debug information that occurs immediately preceeding a Declined message (one or two messages should do the trick).
If you check the cli, the "Declined" message will be right AFTER the sip debug that caused it. So post the sip debug information that occurs immediately preceeding a Declined message (one or two messages should do the trick).
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
- williamconley
- Posts: 20258
- Joined: Wed Oct 31, 2007 4:17 pm
- Location: Davenport, FL (By Disney!)
by sofcall » Sun Jul 03, 2011 6:48 pm
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000028615" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] Audio is at 192.168.1.3 port 10046
[Jul 4 01:43:47] Adding codec 0x100 (g729) to SDP
[Jul 4 01:43:47] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 4 01:43:47] Reliably Transmitting (NAT) to 46.182.3.50:5060:
INVITE sip:90386385208@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK116c27b4;rport
From: "V7040143450000028615" <sip:1870245315@192.168.1.3>;tag=as6c2b1eee
To: <sip:90386385208@46.182.3.50;cpd=on>
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 44329fee4b3087d85407632529d97dc2@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000028615" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Proxy-Authorization: Digest username="1870245315", realm="asterisk", algorithm=MD5, uri="sip:90386385208@46.182.3.50;cpd=on", nonce="36c019ba", response="8d2876981e2ff633204bf093df934be6"
Date: Sun, 03 Jul 2011 23:43:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 2501 2502 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 10046 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK47d4eb15;received=192.168.1.3;rport=49375
From: "V7040143450000023062" <sip:1870245315@192.168.1.3>;tag=as69720335
To: <sip:90386391022@46.182.3.50;cpd=on>
Call-ID: 752c26a37b7e279200b259a935aaf113@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:90386391022@46.182.3.50:5060>
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (11 headers 0 lines) ---
[Jul 4 01:43:47] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK5681b6f2;received=192.168.1.3;rport=49375
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>;tag=as286a35dc
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 102 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="06118b09"
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (11 headers 0 lines) ---
[Jul 4 01:43:47] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90386283421@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK5681b6f2;rport
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>;tag=as286a35dc
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000023810" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] Audio is at 192.168.1.3 port 16594
[Jul 4 01:43:47] Adding codec 0x100 (g729) to SDP
[Jul 4 01:43:47] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 4 01:43:47] Reliably Transmitting (NAT) to 46.182.3.50:5060:
INVITE sip:90386283421@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0b48c537;rport
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000023810" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Proxy-Authorization: Digest username="1870245315", realm="asterisk", algorithm=MD5, uri="sip:90386283421@46.182.3.50;cpd=on", nonce="06118b09", response="257e90ed73b4b11a52f265a0ab0db615"
Date: Sun, 03 Jul 2011 23:43:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 2501 2502 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 16594 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK116c27b4;received=192.168.1.3;rport=49375
From: "V7040143450000028615" <sip:1870245315@192.168.1.3>;tag=as6c2b1eee
To: <sip:90386385208@46.182.3.50;cpd=on>
Call-ID: 44329fee4b3087d85407632529d97dc2@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:90386385208@46.182.3.50:5060>
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (11 headers 0 lines) ---
[Jul 4 01:43:47] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK47d4eb15;received=192.168.1.3;rport=49375
From: "V7040143450000023062" <sip:1870245315@192.168.1.3>;tag=as69720335
To: <sip:90386391022@46.182.3.50;cpd=on>;tag=as5f7cbc6a
Call-ID: 752c26a37b7e279200b259a935aaf113@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (10 headers 0 lines) ---
[Jul 4 01:43:47] -- Got SIP response 603 "Declined" back from 46.182.3.50
[Jul 4 01:43:47] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90386391022@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK47d4eb15;rport
From: "V7040143450000023062" <sip:1870245315@192.168.1.3>;tag=as69720335
To: <sip:90386391022@46.182.3.50;cpd=on>;tag=as5f7cbc6a
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 752c26a37b7e279200b259a935aaf113@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000023062" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] -- SIP/siptrunk-0000015d is busy
[Jul 4 01:43:47] == Everyone is busy/congested at this time (1:1/0/0)
[Jul 4 01:43:47] -- Executing [90386391022@default:3] Hangup("Local/90386391022@default-c747,2", "") in new stack
[Jul 4 01:43:47] == Spawn extension (default, 90386391022, 3) exited non-zero on 'Local/90386391022@default-c747,2'
[Jul 4 01:43:47] -- Executing [h@default:1] DeadAGI("Local/90386391022@default-c747,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Jul 4 01:43:47] Really destroying SIP dialog '752c26a37b7e279200b259a935aaf113@192.168.1.3' Method: INVITE
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0b48c537;received=192.168.1.3;rport=49375
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:90386283421@46.182.3.50:5060>
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (11 headers 0 lines) ---
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK116c27b4;received=192.168.1.3;rport=49375
From: "V7040143450000028615" <sip:1870245315@192.168.1.3>;tag=as6c2b1eee
To: <sip:90386385208@46.182.3.50;cpd=on>;tag=as627c7f92
Call-ID: 44329fee4b3087d85407632529d97dc2@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (10 headers 0 lines) ---
[Jul 4 01:43:47] -- Got SIP response 603 "Declined" back from 46.182.3.50
[Jul 4 01:43:47] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90386385208@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK116c27b4;rport
From: "V7040143450000028615" <sip:1870245315@192.168.1.3>;tag=as6c2b1eee
To: <sip:90386385208@46.182.3.50;cpd=on>;tag=as627c7f92
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 44329fee4b3087d85407632529d97dc2@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000028615" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] -- SIP/siptrunk-0000015e is busy
[Jul 4 01:43:47] == Everyone is busy/congested at this time (1:1/0/0)
[Jul 4 01:43:47] -- Executing [90386385208@default:3] Hangup("Local/90386385208@default-8f06,2", "") in new stack
[Jul 4 01:43:47] == Spawn extension (default, 90386385208, 3) exited non-zero on 'Local/90386385208@default-8f06,2'
[Jul 4 01:43:47] -- Executing [h@default:1] DeadAGI("Local/90386385208@default-8f06,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Jul 4 01:43:47] Really destroying SIP dialog '44329fee4b3087d85407632529d97dc2@192.168.1.3' Method: INVITE
[Jul 4 01:43:47] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0b48c537;received=192.168.1.3;rport=49375
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>;tag=as7f203581
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (10 headers 0 lines) ---
[Jul 4 01:43:47] -- Got SIP response 603 "Declined" back from 46.182.3.50
[Jul 4 01:43:47] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90386283421@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0b48c537;rport
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>;tag=as7f203581
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000023810" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] -- SIP/siptrunk-0000015f is busy
[Jul 4 01:43:47] == Everyone is busy/congested at this time (1:1/0/0)
[Jul 4 01:43:47] -- Executing [90386283421@default:3] Hangup("Local/90386283421@default-036b,2", "") in new stack
[Jul 4 01:43:47] == Spawn extension (default, 90386283421, 3) exited non-zero on 'Local/90386283421@default-036b,2'
[Jul 4 01:43:47] -- Executing [h@default:1] DeadAGI("Local/90386283421@default-036b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Jul 4 01:43:47] Really destroying SIP dialog '5a7047b82846ca5d177731da69c426b1@192.168.1.3' Method: INVITE
[Jul 4 01:43:48] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 4 01:43:48] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 4 01:43:48] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 4 01:43:49] == Parsing '/etc/asterisk/manager.conf': [Jul 4 01:43:49] Found
[Jul 4 01:43:49] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 4 01:43:49] -- Executing [90961320019@default:1] AGI("Local/90961320019@default-9eab,2", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 4 01:43:49] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 4 01:43:49] -- Executing [90961320019@default:2] Dial("Local/90961320019@default-9eab,2", "SIP/siptrunk/90961320019||To") in new stack
[Jul 4 01:43:49] Audio is at 192.168.1.3 port 19316
[Jul 4 01:43:49] Adding codec 0x100 (g729) to SDP
[Jul 4 01:43:49] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 4 01:43:49] Reliably Transmitting (NAT) to 46.182.3.50:5060:
INVITE sip:90961320019@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK20b15478;rport
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143480000027098" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Date: Sun, 03 Jul 2011 23:43:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 2501 2501 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 19316 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jul 4 01:43:49] -- Called siptrunk/90961320019
[Jul 4 01:43:49]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK20b15478;received=192.168.1.3;rport=49375
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>;tag=as423cc2c7
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 102 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="62c14042"
Content-Length: 0
<------------->
[Jul 4 01:43:49] --- (11 headers 0 lines) ---
[Jul 4 01:43:49] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90961320019@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK20b15478;rport
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>;tag=as423cc2c7
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143480000027098" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:49] Audio is at 192.168.1.3 port 19316
[Jul 4 01:43:49] Adding codec 0x100 (g729) to SDP
[Jul 4 01:43:49] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 4 01:43:49] Reliably Transmitting (NAT) to 46.182.3.50:5060:
INVITE sip:90961320019@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0ab73019;rport
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143480000027098" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Proxy-Authorization: Digest username="1870245315", realm="asterisk", algorithm=MD5, uri="sip:90961320019@46.182.3.50;cpd=on", nonce="62c14042", response="ea26a497f745518a695a111bfcb4f8f4"
Date: Sun, 03 Jul 2011 23:43:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 2501 2502 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 19316 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jul 4 01:43:49]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0ab73019;received=192.168.1.3;rport=49375
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:90961320019@46.182.3.50:5060>
Content-Length: 0
<------------->
[Jul 4 01:43:49] --- (11 headers 0 lines) ---
[Jul 4 01:43:49]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0ab73019;received=192.168.1.3;rport=49375
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>;tag=as26a2fb6f
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Jul 4 01:43:49] --- (10 headers 0 lines) ---
[Jul 4 01:43:49] -- Got SIP response 603 "Declined" back from 46.182.3.50
[Jul 4 01:43:49] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90961320019@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0ab73019;rport
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>;tag=as26a2fb6f
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143480000027098" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:49] -- SIP/siptrunk-00000160 is busy
[Jul 4 01:43:49] == Everyone is busy/congested at this time (1:1/0/0)
[Jul 4 01:43:49] -- Executing [90961320019@default:3] Hangup("Local/90961320019@default-9eab,2", "") in new stack
[Jul 4 01:43:49] == Spawn extension (default, 90961320019, 3) exited non-zero on 'Local/90961320019@default-9eab,2'
[Jul 4 01:43:49] -- Executing [h@default:1] DeadAGI("Local/90961320019@default-9eab,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Jul 4 01:43:49] Really destroying SIP dialog '03bf84472931fd295fadc4703790e613@192.168.1.3' Method: INVITE
[Jul 4 01:43:49] == Manager 'sendcron' logged off from 127.0.0.1
go*CLI>
*********************
what would be the cause of that carrier rejecting my calls.
i have a total access to that carrier.46.182......since it's our dedicated server(i m an IT in a callcenter and a minute resseler too. we have many costumers in our server)
Max-Forwards: 70
Remote-Party-ID: "V7040143450000028615" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] Audio is at 192.168.1.3 port 10046
[Jul 4 01:43:47] Adding codec 0x100 (g729) to SDP
[Jul 4 01:43:47] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 4 01:43:47] Reliably Transmitting (NAT) to 46.182.3.50:5060:
INVITE sip:90386385208@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK116c27b4;rport
From: "V7040143450000028615" <sip:1870245315@192.168.1.3>;tag=as6c2b1eee
To: <sip:90386385208@46.182.3.50;cpd=on>
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 44329fee4b3087d85407632529d97dc2@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000028615" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Proxy-Authorization: Digest username="1870245315", realm="asterisk", algorithm=MD5, uri="sip:90386385208@46.182.3.50;cpd=on", nonce="36c019ba", response="8d2876981e2ff633204bf093df934be6"
Date: Sun, 03 Jul 2011 23:43:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 2501 2502 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 10046 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK47d4eb15;received=192.168.1.3;rport=49375
From: "V7040143450000023062" <sip:1870245315@192.168.1.3>;tag=as69720335
To: <sip:90386391022@46.182.3.50;cpd=on>
Call-ID: 752c26a37b7e279200b259a935aaf113@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:90386391022@46.182.3.50:5060>
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (11 headers 0 lines) ---
[Jul 4 01:43:47] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK5681b6f2;received=192.168.1.3;rport=49375
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>;tag=as286a35dc
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 102 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="06118b09"
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (11 headers 0 lines) ---
[Jul 4 01:43:47] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90386283421@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK5681b6f2;rport
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>;tag=as286a35dc
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000023810" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] Audio is at 192.168.1.3 port 16594
[Jul 4 01:43:47] Adding codec 0x100 (g729) to SDP
[Jul 4 01:43:47] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 4 01:43:47] Reliably Transmitting (NAT) to 46.182.3.50:5060:
INVITE sip:90386283421@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0b48c537;rport
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000023810" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Proxy-Authorization: Digest username="1870245315", realm="asterisk", algorithm=MD5, uri="sip:90386283421@46.182.3.50;cpd=on", nonce="06118b09", response="257e90ed73b4b11a52f265a0ab0db615"
Date: Sun, 03 Jul 2011 23:43:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 2501 2502 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 16594 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK116c27b4;received=192.168.1.3;rport=49375
From: "V7040143450000028615" <sip:1870245315@192.168.1.3>;tag=as6c2b1eee
To: <sip:90386385208@46.182.3.50;cpd=on>
Call-ID: 44329fee4b3087d85407632529d97dc2@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:90386385208@46.182.3.50:5060>
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (11 headers 0 lines) ---
[Jul 4 01:43:47] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK47d4eb15;received=192.168.1.3;rport=49375
From: "V7040143450000023062" <sip:1870245315@192.168.1.3>;tag=as69720335
To: <sip:90386391022@46.182.3.50;cpd=on>;tag=as5f7cbc6a
Call-ID: 752c26a37b7e279200b259a935aaf113@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (10 headers 0 lines) ---
[Jul 4 01:43:47] -- Got SIP response 603 "Declined" back from 46.182.3.50
[Jul 4 01:43:47] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90386391022@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK47d4eb15;rport
From: "V7040143450000023062" <sip:1870245315@192.168.1.3>;tag=as69720335
To: <sip:90386391022@46.182.3.50;cpd=on>;tag=as5f7cbc6a
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 752c26a37b7e279200b259a935aaf113@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000023062" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] -- SIP/siptrunk-0000015d is busy
[Jul 4 01:43:47] == Everyone is busy/congested at this time (1:1/0/0)
[Jul 4 01:43:47] -- Executing [90386391022@default:3] Hangup("Local/90386391022@default-c747,2", "") in new stack
[Jul 4 01:43:47] == Spawn extension (default, 90386391022, 3) exited non-zero on 'Local/90386391022@default-c747,2'
[Jul 4 01:43:47] -- Executing [h@default:1] DeadAGI("Local/90386391022@default-c747,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Jul 4 01:43:47] Really destroying SIP dialog '752c26a37b7e279200b259a935aaf113@192.168.1.3' Method: INVITE
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0b48c537;received=192.168.1.3;rport=49375
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:90386283421@46.182.3.50:5060>
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (11 headers 0 lines) ---
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK116c27b4;received=192.168.1.3;rport=49375
From: "V7040143450000028615" <sip:1870245315@192.168.1.3>;tag=as6c2b1eee
To: <sip:90386385208@46.182.3.50;cpd=on>;tag=as627c7f92
Call-ID: 44329fee4b3087d85407632529d97dc2@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (10 headers 0 lines) ---
[Jul 4 01:43:47] -- Got SIP response 603 "Declined" back from 46.182.3.50
[Jul 4 01:43:47] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90386385208@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK116c27b4;rport
From: "V7040143450000028615" <sip:1870245315@192.168.1.3>;tag=as6c2b1eee
To: <sip:90386385208@46.182.3.50;cpd=on>;tag=as627c7f92
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 44329fee4b3087d85407632529d97dc2@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000028615" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] -- SIP/siptrunk-0000015e is busy
[Jul 4 01:43:47] == Everyone is busy/congested at this time (1:1/0/0)
[Jul 4 01:43:47] -- Executing [90386385208@default:3] Hangup("Local/90386385208@default-8f06,2", "") in new stack
[Jul 4 01:43:47] == Spawn extension (default, 90386385208, 3) exited non-zero on 'Local/90386385208@default-8f06,2'
[Jul 4 01:43:47] -- Executing [h@default:1] DeadAGI("Local/90386385208@default-8f06,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Jul 4 01:43:47] Really destroying SIP dialog '44329fee4b3087d85407632529d97dc2@192.168.1.3' Method: INVITE
[Jul 4 01:43:47] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 4 01:43:47]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0b48c537;received=192.168.1.3;rport=49375
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>;tag=as7f203581
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Jul 4 01:43:47] --- (10 headers 0 lines) ---
[Jul 4 01:43:47] -- Got SIP response 603 "Declined" back from 46.182.3.50
[Jul 4 01:43:47] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90386283421@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0b48c537;rport
From: "V7040143450000023810" <sip:1870245315@192.168.1.3>;tag=as24ec769f
To: <sip:90386283421@46.182.3.50;cpd=on>;tag=as7f203581
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 5a7047b82846ca5d177731da69c426b1@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143450000023810" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:47] -- SIP/siptrunk-0000015f is busy
[Jul 4 01:43:47] == Everyone is busy/congested at this time (1:1/0/0)
[Jul 4 01:43:47] -- Executing [90386283421@default:3] Hangup("Local/90386283421@default-036b,2", "") in new stack
[Jul 4 01:43:47] == Spawn extension (default, 90386283421, 3) exited non-zero on 'Local/90386283421@default-036b,2'
[Jul 4 01:43:47] -- Executing [h@default:1] DeadAGI("Local/90386283421@default-036b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Jul 4 01:43:47] Really destroying SIP dialog '5a7047b82846ca5d177731da69c426b1@192.168.1.3' Method: INVITE
[Jul 4 01:43:48] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 4 01:43:48] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 4 01:43:48] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 4 01:43:49] == Parsing '/etc/asterisk/manager.conf': [Jul 4 01:43:49] Found
[Jul 4 01:43:49] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 4 01:43:49] -- Executing [90961320019@default:1] AGI("Local/90961320019@default-9eab,2", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 4 01:43:49] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 4 01:43:49] -- Executing [90961320019@default:2] Dial("Local/90961320019@default-9eab,2", "SIP/siptrunk/90961320019||To") in new stack
[Jul 4 01:43:49] Audio is at 192.168.1.3 port 19316
[Jul 4 01:43:49] Adding codec 0x100 (g729) to SDP
[Jul 4 01:43:49] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 4 01:43:49] Reliably Transmitting (NAT) to 46.182.3.50:5060:
INVITE sip:90961320019@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK20b15478;rport
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143480000027098" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Date: Sun, 03 Jul 2011 23:43:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 2501 2501 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 19316 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jul 4 01:43:49] -- Called siptrunk/90961320019
[Jul 4 01:43:49]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK20b15478;received=192.168.1.3;rport=49375
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>;tag=as423cc2c7
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 102 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="62c14042"
Content-Length: 0
<------------->
[Jul 4 01:43:49] --- (11 headers 0 lines) ---
[Jul 4 01:43:49] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90961320019@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK20b15478;rport
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>;tag=as423cc2c7
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143480000027098" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:49] Audio is at 192.168.1.3 port 19316
[Jul 4 01:43:49] Adding codec 0x100 (g729) to SDP
[Jul 4 01:43:49] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 4 01:43:49] Reliably Transmitting (NAT) to 46.182.3.50:5060:
INVITE sip:90961320019@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0ab73019;rport
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143480000027098" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Proxy-Authorization: Digest username="1870245315", realm="asterisk", algorithm=MD5, uri="sip:90961320019@46.182.3.50;cpd=on", nonce="62c14042", response="ea26a497f745518a695a111bfcb4f8f4"
Date: Sun, 03 Jul 2011 23:43:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 2501 2502 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 19316 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jul 4 01:43:49]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0ab73019;received=192.168.1.3;rport=49375
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:90961320019@46.182.3.50:5060>
Content-Length: 0
<------------->
[Jul 4 01:43:49] --- (11 headers 0 lines) ---
[Jul 4 01:43:49]
<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0ab73019;received=192.168.1.3;rport=49375
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>;tag=as26a2fb6f
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Jul 4 01:43:49] --- (10 headers 0 lines) ---
[Jul 4 01:43:49] -- Got SIP response 603 "Declined" back from 46.182.3.50
[Jul 4 01:43:49] Transmitting (NAT) to 46.182.3.50:5060:
ACK sip:90961320019@46.182.3.50;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK0ab73019;rport
From: "V7040143480000027098" <sip:1870245315@192.168.1.3>;tag=as099afb4c
To: <sip:90961320019@46.182.3.50;cpd=on>;tag=as26a2fb6f
Contact: <sip:1870245315@192.168.1.3>
Call-ID: 03bf84472931fd295fadc4703790e613@192.168.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V7040143480000027098" <sip:0000000000@192.168.1.3>;privacy=off;screen=no
Content-Length: 0
---
[Jul 4 01:43:49] -- SIP/siptrunk-00000160 is busy
[Jul 4 01:43:49] == Everyone is busy/congested at this time (1:1/0/0)
[Jul 4 01:43:49] -- Executing [90961320019@default:3] Hangup("Local/90961320019@default-9eab,2", "") in new stack
[Jul 4 01:43:49] == Spawn extension (default, 90961320019, 3) exited non-zero on 'Local/90961320019@default-9eab,2'
[Jul 4 01:43:49] -- Executing [h@default:1] DeadAGI("Local/90961320019@default-9eab,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Jul 4 01:43:49] Really destroying SIP dialog '03bf84472931fd295fadc4703790e613@192.168.1.3' Method: INVITE
[Jul 4 01:43:49] == Manager 'sendcron' logged off from 127.0.0.1
go*CLI>
*********************
what would be the cause of that carrier rejecting my calls.
i have a total access to that carrier.46.182......since it's our dedicated server(i m an IT in a callcenter and a minute resseler too. we have many costumers in our server)
- sofcall
- Posts: 110
- Joined: Wed Feb 02, 2011 4:49 am
- Location: Morocco
by williamconley » Sun Jul 03, 2011 10:31 pm
sorry, but no new information. you'll have to check with your sip provider why they are declining your call.<--- SIP read from 46.182.3.50:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK47d4eb15;received=192.168.1.3;rport=49375
From: "V7040143450000023062" <sip:1870245315@192.168.1.3>;tag=as69720335
To: <sip:90386391022@46.182.3.50;cpd=on>;tag=as5f7cbc6a
Call-ID: 752c26a37b7e279200b259a935aaf113@192.168.1.3
CSeq: 103 INVITE
although it is possible they do not like your ip address being listed as 192.168.1.3, you may wish to modify your "externip=" command (or add it or uncomment it) in sip.conf and enter your external ip address in the dialog (and perhaps set your NAT for this carrier to "nat=yes".
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
- williamconley
- Posts: 20258
- Joined: Wed Oct 31, 2007 4:17 pm
- Location: Davenport, FL (By Disney!)
outbound dialling
by striker » Sun Jul 03, 2011 11:41 pm
Hi
1. register your sip trunk settings in a softphone , and make sure that you are able to dial the Numbers from the softphone.
2. as i have seen that your sip trunk needs G729 codec, is the g729 is installed in ur server , and r u using a g729 enabled softphone?
3. type show transalation in the asterisk cli , it will show wether g729 in loaded or not.
4. also check sip show peers and sip show registry in cli
1. register your sip trunk settings in a softphone , and make sure that you are able to dial the Numbers from the softphone.
2. as i have seen that your sip trunk needs G729 codec, is the g729 is installed in ur server , and r u using a g729 enabled softphone?
3. type show transalation in the asterisk cli , it will show wether g729 in loaded or not.
4. also check sip show peers and sip show registry in cli
www.striker24x7.com www.youtube.com/c/striker24x7 Telegram/skype id : striker24x7
- striker
- Posts: 962
- Joined: Sun Jun 06, 2010 10:25 am
by sofcall » Mon Jul 04, 2011 2:57 pm
Comming back home.did no modification and it works great.i mean PREDECTIVE DIALING.
Also tried manuel just to be sure but that s what happens:
i can make calls but without hearing the phone ringing(i hear nothing in the otherside till i hear the costumer saying hello) and then when i statue the call the button DIAL NEXT get inactive and i had to desconnect and connect again to receive a call.
cli when making manuel dial methode:
Connected to Asterisk 1.4.39.1-vici RPM by demian@goautodial.com currently running on go (pid = 8610)
Verbosity is at least 3
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8600051@default:1] MeetMe("Local/8600051@default-e6d9,2", "8600051|F") in new stack
-- Executing [0386577318@default:1] AGI("Local/8600051@default-e6d9,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [0386577318@default:2] Dial("Local/8600051@default-e6d9,1", "SIP/siptrunk/0386577318||To") in new stack
-- Called siptrunk/0386577318
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/siptrunk-0000002a is ringing
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/siptrunk-0000002a answered Local/8600051@default-e6d9,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [h@default:1] DeadAGI("Local/8600051@default-e6d9,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----37-----5") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --37-----5 completed, returning 0
== Spawn extension (default, 0386577318, 2) exited non-zero on 'Local/8600051@default-e6d9,1'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-e6d9,2'
-- Executing [h@default:1] DeadAGI("Local/8600051@default-e6d9,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
***********************
sip show peers as you asked:
go*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
cc102/cc102 192.168.1.74 D N 17810 OK (112 ms)
8020/8020 (Unspecified) D N 0 UNKNOWN
8019/8019 (Unspecified) D N 0 UNKNOWN
*********************
sip show translation
Connected to Asterisk 1.4.39.1-vici RPM by demian@goautodial.com currently running on go (pid = 8610)
Verbosity is at least 3
go*CLI> core show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 4 4 8 4 3 9 16 - - 8 -
ulaw - 6 - 1 6 2 1 7 14 - - 6 -
alaw - 6 1 - 6 2 1 7 14 - - 6 -
g726aal2 - 9 5 5 - 5 4 10 17 - - 9 -
adpcm - 6 2 2 6 - 1 7 14 - - 6 -
slin - 5 1 1 5 1 - 6 13 - - 5 -
lpc10 - 9 5 5 9 5 4 - 17 - - 9 -
g729 - 9 5 5 9 5 4 10 - - - 9 -
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 - 9 5 5 9 5 4 10 17 - - - -
g722 - - - - - - - - - - - - -
go*CLI>
------------------------
sip show registry:
go*CLI> sip show registry
Host Username Refresh State Reg.Time
46.182.3.50:5060 1870245315 105 Registered Mon,
****************************
By the way im already using.nat=yes
thank you very much.
Also tried manuel just to be sure but that s what happens:
i can make calls but without hearing the phone ringing(i hear nothing in the otherside till i hear the costumer saying hello) and then when i statue the call the button DIAL NEXT get inactive and i had to desconnect and connect again to receive a call.
cli when making manuel dial methode:
Connected to Asterisk 1.4.39.1-vici RPM by demian@goautodial.com currently running on go (pid = 8610)
Verbosity is at least 3
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8600051@default:1] MeetMe("Local/8600051@default-e6d9,2", "8600051|F") in new stack
-- Executing [0386577318@default:1] AGI("Local/8600051@default-e6d9,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [0386577318@default:2] Dial("Local/8600051@default-e6d9,1", "SIP/siptrunk/0386577318||To") in new stack
-- Called siptrunk/0386577318
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/siptrunk-0000002a is ringing
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/siptrunk-0000002a answered Local/8600051@default-e6d9,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [h@default:1] DeadAGI("Local/8600051@default-e6d9,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----37-----5") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --37-----5 completed, returning 0
== Spawn extension (default, 0386577318, 2) exited non-zero on 'Local/8600051@default-e6d9,1'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-e6d9,2'
-- Executing [h@default:1] DeadAGI("Local/8600051@default-e6d9,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
***********************
sip show peers as you asked:
go*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
cc102/cc102 192.168.1.74 D N 17810 OK (112 ms)
8020/8020 (Unspecified) D N 0 UNKNOWN
8019/8019 (Unspecified) D N 0 UNKNOWN
*********************
sip show translation
Connected to Asterisk 1.4.39.1-vici RPM by demian@goautodial.com currently running on go (pid = 8610)
Verbosity is at least 3
go*CLI> core show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 4 4 8 4 3 9 16 - - 8 -
ulaw - 6 - 1 6 2 1 7 14 - - 6 -
alaw - 6 1 - 6 2 1 7 14 - - 6 -
g726aal2 - 9 5 5 - 5 4 10 17 - - 9 -
adpcm - 6 2 2 6 - 1 7 14 - - 6 -
slin - 5 1 1 5 1 - 6 13 - - 5 -
lpc10 - 9 5 5 9 5 4 - 17 - - 9 -
g729 - 9 5 5 9 5 4 10 - - - 9 -
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 - 9 5 5 9 5 4 10 17 - - - -
g722 - - - - - - - - - - - - -
go*CLI>
------------------------
sip show registry:
go*CLI> sip show registry
Host Username Refresh State Reg.Time
46.182.3.50:5060 1870245315 105 Registered Mon,
****************************
By the way im already using.nat=yes
thank you very much.
- sofcall
- Posts: 110
- Joined: Wed Feb 02, 2011 4:49 am
- Location: Morocco
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