Hi, I am having issues with my dialer once more and below is the problem that shows in the asterisk -r screen. I wanted to ask if the problem is on my end or the hosts end. The reason I ask is because I guess my is so perfect that they never have any problems at all what so ever and whatever problem occurs it is always on my end. Then I have to spend lots of money hiring someone to fix something that is never on my end.
Below I have the sip debug as well to see what you can find in between their also. I apologize is the post is extremely long.
-- Executing AGI("SIP/1010-b7b15be0", "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 Dial("SIP/1010-b7b15be0", "SIP/eccomm/13056422927|60|tTo") in new stack
-- Called eccomm/13056422927
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
-- SIP/eccomm-09d7aed8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("SIP/1010-b7b15be0", "") in new stack
== Spawn extension (default, 813056422927, 3) exited non-zero on 'SIP/1010-b7b15be0'
-- Executing DeadAGI("SIP/1010-b7b15be0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Welcome to VICIDIALNOW!!!
-------------------------------------------------
For access to the VICIDIAL admin and agent web GUI use this URL:
http://172.17.3.85
username: admin
password: vicidialnow
For access to VtigerCRM use this URL:
http://172.17.3.85/vtigercrm
username: admin
password: admin
For professional support, visit http://www.vicidialnow.com
-------------------------------------------------
Don't forget to run update_server_ip everytime you change your IP address
]0;root@dialdatatech:~[root@dialdatatech ~]# asterisk -r
Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.27 currently running on dialdatatech (pid = 2262)
dialdatatech*CLI>
Verbosity is at least 21
[Kdialdatatech*CLI> sip no debug
dialdatatech*CLI>
SIP Debugging Disabled
[Kdialdatatech*CLI> sip debug
dialdatatech*CLI>
SIP Debugging enabled
[Kdialdatatech*CLI>
12 headers, 0 lines
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
OPTIONS sip:63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK4ccf25d9;rport
From: "asterisk" <sip:asterisk@190.10.19.79>;tag=as067fd7ab
To: <sip:63.251.216.50>
Contact: <sip:asterisk@190.10.19.79>
Call-ID: 6ecd23962bcfecdb65e8f71d150808f6@190.10.19.79
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK4ccf25d9;rport=7390
From: "asterisk" <sip:asterisk@190.10.19.79>;tag=as067fd7ab
To: <sip:63.251.216.50>;tag=xcast-2969777400
Call-ID: 6ecd23962bcfecdb65e8f71d150808f6@190.10.19.79
CSeq: 102 OPTIONS
Server: XCast Carrier/1.0
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
Destroying call '6ecd23962bcfecdb65e8f71d150808f6@190.10.19.79'
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/817063648093@default-8508,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/817063648093@default-8508,2[0;37;40m", "[1;35;40mSIP/eccomm/17063648093|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 18468
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:17063648093@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5629a48a;rport
From: "V1125095604000091893" <sip:4018291847@190.10.19.79>;tag=as46dbff38
To: <sip:17063648093@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 6ce95a4d11b6f1792b48d94571bdc096@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 18468 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/17063648093
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5629a48a;rport=7390
From: "V1125095604000091893" <sip:4018291847@190.10.19.79>;tag=as46dbff38
To: <sip:17063648093@63.251.216.50>
Call-ID: 6ce95a4d11b6f1792b48d94571bdc096@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
[Kdialdatatech*CLI>
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5629a48a;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095604000091893" <sip:4018291847@190.10.19.79>;tag=as46dbff38
To: <sip:17063648093@63.251.216.50>;tag=xcast-257982112
Call-ID: 6ce95a4d11b6f1792b48d94571bdc096@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
[Kdialdatatech*CLI>
--- (9 headers 0 lines) ---
[Kdialdatatech*CLI>
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
[Kdialdatatech*CLI>
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:17063648093@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5629a48a;rport
From: "V1125095604000091893" <sip:4018291847@190.10.19.79>;tag=as46dbff38
To: <sip:17063648093@63.251.216.50>;tag=xcast-257982112
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 6ce95a4d11b6f1792b48d94571bdc096@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d8af90 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/817063648093@default-8508,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 817063648093, 3) exited non-zero on 'Local/817063648093@default-8508,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/817063648093@default-8508,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
Destroying call '6ce95a4d11b6f1792b48d94571bdc096@190.10.19.79'
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/817708268608@default-0128,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/817708268608@default-0128,2[0;37;40m", "[1;35;40mSIP/eccomm/17708268608|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 10388
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:17708268608@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK2da4861b;rport
From: "V1125095604000091887" <sip:4018291847@190.10.19.79>;tag=as797e8c89
To: <sip:17708268608@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 30bafe7d094428a90e01873c21c7b9db@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 10388 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/17708268608
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/814043647684@default-bb86,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/814043647684@default-bb86,2[0;37;40m", "[1;35;40mSIP/eccomm/14043647684|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 13034
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:14043647684@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5e606f5d;rport
From: "V1125095604000091883" <sip:4018291847@190.10.19.79>;tag=as74b2c936
To: <sip:14043647684@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 71f5ced56b4705f83ad02cab4f7b35d2@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 13034 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/14043647684
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/812392972883@default-1e45,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/812392972883@default-1e45,2[0;37;40m", "[1;35;40mSIP/eccomm/12392972883|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 19760
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:12392972883@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK1a5ab279;rport
From: "V1125095604000091875" <sip:4018291847@190.10.19.79>;tag=as04bc0aa6
To: <sip:12392972883@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 391d650a4104f4b33a1424bc28e14852@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 19760 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/12392972883
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK2da4861b;rport=7390
From: "V1125095604000091887" <sip:4018291847@190.10.19.79>;tag=as797e8c89
To: <sip:17708268608@63.251.216.50>
Call-ID: 30bafe7d094428a90e01873c21c7b9db@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK2da4861b;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095604000091887" <sip:4018291847@190.10.19.79>;tag=as797e8c89
To: <sip:17708268608@63.251.216.50>;tag=xcast-46912522222480
Call-ID: 30bafe7d094428a90e01873c21c7b9db@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:17708268608@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK2da4861b;rport
From: "V1125095604000091887" <sip:4018291847@190.10.19.79>;tag=as797e8c89
To: <sip:17708268608@63.251.216.50>;tag=xcast-46912522222480
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 30bafe7d094428a90e01873c21c7b9db@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
[Kdialdatatech*CLI>
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d8af90 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/817708268608@default-0128,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 817708268608, 3) exited non-zero on 'Local/817708268608@default-0128,2'
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/817708268608@default-0128,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
Destroying call '30bafe7d094428a90e01873c21c7b9db@190.10.19.79'
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5e606f5d;rport=7390
From: "V1125095604000091883" <sip:4018291847@190.10.19.79>;tag=as74b2c936
To: <sip:14043647684@63.251.216.50>
Call-ID: 71f5ced56b4705f83ad02cab4f7b35d2@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5e606f5d;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095604000091883" <sip:4018291847@190.10.19.79>;tag=as74b2c936
To: <sip:14043647684@63.251.216.50>;tag=xcast-2954001072
Call-ID: 71f5ced56b4705f83ad02cab4f7b35d2@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:14043647684@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5e606f5d;rport
From: "V1125095604000091883" <sip:4018291847@190.10.19.79>;tag=as74b2c936
To: <sip:14043647684@63.251.216.50>;tag=xcast-2954001072
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 71f5ced56b4705f83ad02cab4f7b35d2@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d9b988 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/814043647684@default-bb86,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 814043647684, 3) exited non-zero on 'Local/814043647684@default-bb86,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/814043647684@default-bb86,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
Destroying call '71f5ced56b4705f83ad02cab4f7b35d2@190.10.19.79'
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK1a5ab279;rport=7390
From: "V1125095604000091875" <sip:4018291847@190.10.19.79>;tag=as04bc0aa6
To: <sip:12392972883@63.251.216.50>
Call-ID: 391d650a4104f4b33a1424bc28e14852@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK1a5ab279;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095604000091875" <sip:4018291847@190.10.19.79>;tag=as04bc0aa6
To: <sip:12392972883@63.251.216.50>;tag=xcast-2976204880
Call-ID: 391d650a4104f4b33a1424bc28e14852@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:12392972883@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK1a5ab279;rport
From: "V1125095604000091875" <sip:4018291847@190.10.19.79>;tag=as04bc0aa6
To: <sip:12392972883@63.251.216.50>;tag=xcast-2976204880
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 391d650a4104f4b33a1424bc28e14852@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09dab1a8 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/812392972883@default-1e45,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 812392972883, 3) exited non-zero on 'Local/812392972883@default-1e45,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/812392972883@default-1e45,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
Destroying call '391d650a4104f4b33a1424bc28e14852@190.10.19.79'
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
12 headers, 0 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 172.17.3.111:42040:
OPTIONS sip:1001@172.17.3.111:42040;rinstance=31f0da1f1e1c0fe3 SIP/2.0
Via: SIP/2.0/UDP 172.17.3.85:5060;branch=z9hG4bK4fa431ce;rport
From: "asterisk" <sip:asterisk@172.17.3.85>;tag=as4fa23541
To: <sip:1001@172.17.3.111:42040;rinstance=31f0da1f1e1c0fe3>
Contact: <sip:asterisk@172.17.3.85>
Call-ID: 797305707d8aca303a0937745df2d3c1@172.17.3.85
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kdialdatatech*CLI>
<-- SIP read from 172.17.3.111:42040:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.3.85:5060;branch=z9hG4bK4fa431ce;rport=5060
Contact: <sip:172.17.3.111:42040>
To: <sip:1001@172.17.3.111:42040;rinstance=31f0da1f1e1c0fe3>;tag=3a6c7c4f
From: "asterisk"<sip:asterisk@172.17.3.85>;tag=as4fa23541
Call-ID: 797305707d8aca303a0937745df2d3c1@172.17.3.85
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1100z stamp 47739
Content-Length: 0
--- (12 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 172.17.3.111:42040:
[Kdialdatatech*CLI>
--- (0 headers 1 lines) ---
[Kdialdatatech*CLI>
Destroying call '797305707d8aca303a0937745df2d3c1@172.17.3.85'
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/814137865347@default-90ab,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/814137865347@default-90ab,2[0;37;40m", "[1;35;40mSIP/eccomm/14137865347|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 12842
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:14137865347@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK11b5152d;rport
From: "V1125095609000091873" <sip:4018291847@190.10.19.79>;tag=as48c0b587
To: <sip:14137865347@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 53cdcaa41d1b81d85c4b33ec6700e46d@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 12842 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/14137865347
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/813172246340@default-e11b,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/813172246340@default-e11b,2[0;37;40m", "[1;35;40mSIP/eccomm/13172246340|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 15798
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:13172246340@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK53b6a7fb;rport
From: "V1125095609000091867" <sip:4018291847@190.10.19.79>;tag=as685d83f6
To: <sip:13172246340@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 3aef4533731355592c79e61408b9bcae@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 15798 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/13172246340
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK11b5152d;rport=7390
From: "V1125095609000091873" <sip:4018291847@190.10.19.79>;tag=as48c0b587
To: <sip:14137865347@63.251.216.50>
Call-ID: 53cdcaa41d1b81d85c4b33ec6700e46d@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK11b5152d;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095609000091873" <sip:4018291847@190.10.19.79>;tag=as48c0b587
To: <sip:14137865347@63.251.216.50>;tag=xcast-2977096888
Call-ID: 53cdcaa41d1b81d85c4b33ec6700e46d@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:14137865347@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK11b5152d;rport
From: "V1125095609000091873" <sip:4018291847@190.10.19.79>;tag=as48c0b587
To: <sip:14137865347@63.251.216.50>;tag=xcast-2977096888
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 53cdcaa41d1b81d85c4b33ec6700e46d@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d8af90 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/814137865347@default-90ab,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 814137865347, 3) exited non-zero on 'Local/814137865347@default-90ab,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/814137865347@default-90ab,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
Destroying call '53cdcaa41d1b81d85c4b33ec6700e46d@190.10.19.79'
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/819176739053@default-3444,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/819176739053@default-3444,2[0;37;40m", "[1;35;40mSIP/eccomm/19176739053|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 15292
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:19176739053@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK4b137f32;rport
From: "V1125095609000091860" <sip:4018291847@190.10.19.79>;tag=as5bcd6de6
To: <sip:19176739053@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 11738a434ace0fbd0531914f379957fb@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 15292 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK53b6a7fb;rport=7390
From: "V1125095609000091867" <sip:4018291847@190.10.19.79>;tag=as685d83f6
To: <sip:13172246340@63.251.216.50>
Call-ID: 3aef4533731355592c79e61408b9bcae@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
-- Called eccomm/19176739053
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK53b6a7fb;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095609000091867" <sip:4018291847@190.10.19.79>;tag=as685d83f6
To: <sip:13172246340@63.251.216.50>;tag=xcast-166830376
Call-ID: 3aef4533731355592c79e61408b9bcae@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:13172246340@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK53b6a7fb;rport
From: "V1125095609000091867" <sip:4018291847@190.10.19.79>;tag=as685d83f6
To: <sip:13172246340@63.251.216.50>;tag=xcast-166830376
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 3aef4533731355592c79e61408b9bcae@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d9b988 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/813172246340@default-e11b,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 813172246340, 3) exited non-zero on 'Local/813172246340@default-e11b,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/813172246340@default-e11b,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/819546464426@default-3ff7,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/819546464426@default-3ff7,2[0;37;40m", "[1;35;40mSIP/eccomm/19546464426|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
Destroying call '3aef4533731355592c79e61408b9bcae@190.10.19.79'
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 12918
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:19546464426@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK587daaf9;rport
From: "V1125095609000091866" <sip:4018291847@190.10.19.79>;tag=as0e515585
To: <sip:19546464426@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 7f5c06d12768493e16508ae30fc39053@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 12918 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/19546464426
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK4b137f32;rport=7390
From: "V1125095609000091860" <sip:4018291847@190.10.19.79>;tag=as5bcd6de6
To: <sip:19176739053@63.251.216.50>
Call-ID: 11738a434ace0fbd0531914f379957fb@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK4b137f32;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095609000091860" <sip:4018291847@190.10.19.79>;tag=as5bcd6de6
To: <sip:19176739053@63.251.216.50>;tag=xcast-2967379808
Call-ID: 11738a434ace0fbd0531914f379957fb@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:19176739053@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK4b137f32;rport
From: "V1125095609000091860" <sip:4018291847@190.10.19.79>;tag=as5bcd6de6
To: <sip:19176739053@63.251.216.50>;tag=xcast-2967379808
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 11738a434ace0fbd0531914f379957fb@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/eccomm-09d8af90 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/819176739053@default-3444,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 819176739053, 3) exited non-zero on 'Local/819176739053@default-3444,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/819176739053@default-3444,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
Destroying call '11738a434ace0fbd0531914f379957fb@190.10.19.79'
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK587daaf9;rport=7390
From: "V1125095609000091866" <sip:4018291847@190.10.19.79>;tag=as0e515585
To: <sip:19546464426@63.251.216.50>
Call-ID: 7f5c06d12768493e16508ae30fc39053@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
[Kdialdatatech*CLI>
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK587daaf9;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095609000091866" <sip:4018291847@190.10.19.79>;tag=as0e515585
To: <sip:19546464426@63.251.216.50>;tag=xcast-46912590636816
Call-ID: 7f5c06d12768493e16508ae30fc39053@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
[Kdialdatatech*CLI>
--- (9 headers 0 lines) ---
[Kdialdatatech*CLI>
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
[Kdialdatatech*CLI>
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:19546464426@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK587daaf9;rport
From: "V1125095609000091866" <sip:4018291847@190.10.19.79>;tag=as0e515585
To: <sip:19546464426@63.251.216.50>;tag=xcast-46912590636816
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 7f5c06d12768493e16508ae30fc39053@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d9b988 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/819546464426@default-3ff7,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 819546464426, 3) exited non-zero on 'Local/819546464426@default-3ff7,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/819546464426@default-3ff7,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
Destroying call '7f5c06d12768493e16508ae30fc39053@190.10.19.79'
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
[Kdialdatatech*CLI>
12 headers, 0 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 172.17.3.104:57778:
OPTIONS sip:1010@172.17.3.104:57778;rinstance=495d1dc3bc9b061f SIP/2.0
Via: SIP/2.0/UDP 172.17.3.85:5060;branch=z9hG4bK06798c37;rport
From: "asterisk" <sip:asterisk@172.17.3.85>;tag=as254a65b9
To: <sip:1010@172.17.3.104:57778;rinstance=495d1dc3bc9b061f>
Contact: <sip:asterisk@172.17.3.85>
Call-ID: 011ccbaa06dd4bef7e5b8cca218ad4b1@172.17.3.85
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
[Kdialdatatech*CLI>
<-- SIP read from 172.17.3.104:57778:
[Kdialdatatech*CLI>
--- (0 headers 1 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 172.17.3.104:57778:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.3.85:5060;branch=z9hG4bK06798c37;rport=5060
Contact: <sip:172.17.3.104:57778>
To: <sip:1010@172.17.3.104:57778;rinstance=495d1dc3bc9b061f>;tag=41041c02
From: "asterisk"<sip:asterisk@172.17.3.85>;tag=as254a65b9
Call-ID: 011ccbaa06dd4bef7e5b8cca218ad4b1@172.17.3.85
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 1011s stamp 41150
Content-Length: 0
--- (12 headers 0 lines) ---
[Kdialdatatech*CLI>
Destroying call '011ccbaa06dd4bef7e5b8cca218ad4b1@172.17.3.85'
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/813027314220@default-1a66,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/813027314220@default-1a66,2[0;37;40m", "[1;35;40mSIP/eccomm/13027314220|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 13164
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:13027314220@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK7595b2d0;rport
From: "V1125095614000091853" <sip:4018291847@190.10.19.79>;tag=as3c066e3a
To: <sip:13027314220@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 522bedaa51cf2185733a0f3b2f2b327e@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 13164 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/13027314220
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/817172482190@default-19c2,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/817172482190@default-19c2,2[0;37;40m", "[1;35;40mSIP/eccomm/17172482190|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 13074
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:17172482190@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK376c3b52;rport
From: "V1125095614000091852" <sip:4018291847@190.10.19.79>;tag=as318730a7
To: <sip:17172482190@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 32770060345ba0ea05bb774735159491@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 13074 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/17172482190
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK7595b2d0;rport=7390
From: "V1125095614000091853" <sip:4018291847@190.10.19.79>;tag=as3c066e3a
To: <sip:13027314220@63.251.216.50>
Call-ID: 522bedaa51cf2185733a0f3b2f2b327e@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK7595b2d0;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095614000091853" <sip:4018291847@190.10.19.79>;tag=as3c066e3a
To: <sip:13027314220@63.251.216.50>;tag=xcast-46912519552288
Call-ID: 522bedaa51cf2185733a0f3b2f2b327e@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:13027314220@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK7595b2d0;rport
From: "V1125095614000091853" <sip:4018291847@190.10.19.79>;tag=as3c066e3a
To: <sip:13027314220@63.251.216.50>;tag=xcast-46912519552288
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 522bedaa51cf2185733a0f3b2f2b327e@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
[Kdialdatatech*CLI>
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d8af90 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/813027314220@default-1a66,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 813027314220, 3) exited non-zero on 'Local/813027314220@default-1a66,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/813027314220@default-1a66,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
Destroying call '522bedaa51cf2185733a0f3b2f2b327e@190.10.19.79'
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK376c3b52;rport=7390
From: "V1125095614000091852" <sip:4018291847@190.10.19.79>;tag=as318730a7
To: <sip:17172482190@63.251.216.50>
Call-ID: 32770060345ba0ea05bb774735159491@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
[Kdialdatatech*CLI>
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK376c3b52;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095614000091852" <sip:4018291847@190.10.19.79>;tag=as318730a7
To: <sip:17172482190@63.251.216.50>;tag=xcast-2960238664
Call-ID: 32770060345ba0ea05bb774735159491@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
[Kdialdatatech*CLI>
--- (9 headers 0 lines) ---
[Kdialdatatech*CLI>
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
[Kdialdatatech*CLI>
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:17172482190@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK376c3b52;rport
From: "V1125095614000091852" <sip:4018291847@190.10.19.79>;tag=as318730a7
To: <sip:17172482190@63.251.216.50>;tag=xcast-2960238664
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 32770060345ba0ea05bb774735159491@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d9b988 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/817172482190@default-19c2,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 817172482190, 3) exited non-zero on 'Local/817172482190@default-19c2,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/817172482190@default-19c2,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/815086682596@default-9308,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/815086682596@default-9308,2[0;37;40m", "[1;35;40mSIP/eccomm/15086682596|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
Destroying call '32770060345ba0ea05bb774735159491@190.10.19.79'
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 14512
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:15086682596@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK7b99df35;rport
From: "V1125095614000091851" <sip:4018291847@190.10.19.79>;tag=as0fceead9
To: <sip:15086682596@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 017894330ae303273d2eba6e442ba6e3@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 14512 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/15086682596
[Kdialdatatech*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
[Kdialdatatech*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
[Kdialdatatech*CLI>
-- Executing [1;36;40mAGI[0;37;40m("[1;35;40mLocal/812034606600@default-0da5,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Kdialdatatech*CLI>
-- Executing [1;36;40mDial[0;37;40m("[1;35;40mLocal/812034606600@default-0da5,2[0;37;40m", "[1;35;40mSIP/eccomm/12034606600|60|tTo[0;37;40m") in new stack
[Kdialdatatech*CLI>
We're at 190.10.19.79 port 12416
[Kdialdatatech*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kdialdatatech*CLI>
Adding codec 0x1 (g723) to SDP
[Kdialdatatech*CLI>
Adding codec 0x100 (g729) to SDP
[Kdialdatatech*CLI>
Adding codec 0x2 (gsm) to SDP
[Kdialdatatech*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kdialdatatech*CLI>
13 headers, 15 lines
[Kdialdatatech*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:12034606600@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK30220019;rport
From: "V1125095614000091849" <sip:4018291847@190.10.19.79>;tag=as6b407c5c
To: <sip:12034606600@63.251.216.50>
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 4f42ca2d2d27395053cc5dfa54e929a4@190.10.19.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 25 Nov 2008 14:56:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2262 2262 IN IP4 190.10.19.79
s=session
c=IN IP4 190.10.19.79
t=0 0
m=audio 12416 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Kdialdatatech*CLI>
-- Called eccomm/12034606600
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK7b99df35;rport=7390
From: "V1125095614000091851" <sip:4018291847@190.10.19.79>;tag=as0fceead9
To: <sip:15086682596@63.251.216.50>
Call-ID: 017894330ae303273d2eba6e442ba6e3@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK7b99df35;rport=7390
Record-Route: <sip:63.251.216.50:5060;lr>
From: "V1125095614000091851" <sip:4018291847@190.10.19.79>;tag=as0fceead9
To: <sip:15086682596@63.251.216.50>;tag=xcast-46912522299712
Call-ID: 017894330ae303273d2eba6e442ba6e3@190.10.19.79
CSeq: 102 INVITE
Server: XCast Carrier/1.0
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 410 "Unknown/Disabled User" back from 63.251.216.50
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:15086682596@63.251.216.50 SIP/2.0
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK7b99df35;rport
From: "V1125095614000091851" <sip:4018291847@190.10.19.79>;tag=as0fceead9
To: <sip:15086682596@63.251.216.50>;tag=xcast-46912522299712
Contact: <sip:4018291847@190.10.19.79>
Call-ID: 017894330ae303273d2eba6e442ba6e3@190.10.19.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
[Kdialdatatech*CLI>
---
[Kdialdatatech*CLI>
-- SIP/eccomm-09d8af90 is circuit-busy
[Kdialdatatech*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
[Kdialdatatech*CLI>
-- Executing [1;36;40mHangup[0;37;40m("[1;35;40mLocal/815086682596@default-9308,2[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Kdialdatatech*CLI>
== Spawn extension (default, 815086682596, 3) exited non-zero on 'Local/815086682596@default-9308,2'
[Kdialdatatech*CLI>
-- Executing [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/815086682596@default-9308,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------[0;37;40m") in new stack
[Kdialdatatech*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Kdialdatatech*CLI>
Destroying call '017894330ae303273d2eba6e442ba6e3@190.10.19.79'
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK30220019;rport=7390
From: "V1125095614000091849" <sip:4018291847@190.10.19.79>;tag=as6b407c5c
To: <sip:12034606600@63.251.216.50>
Call-ID: 4f42ca2d2d27395053cc5dfa54e929a4@190.10.19.79
CSeq: 102 INVITE
Server: XCast Partner/1.1
Content-Length: 0
[Kdialdatatech*CLI>
--- (8 headers 0 lines) ---
[Kdialdatatech*CLI>
<-- SIP read from 63.251.216.50:5060:
SIP/2.0 410 Unknown/Disabled User
Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK30220019;rport=7390
Record-Route: <sip:63.251.216.50
having sip response 410 issue
Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N
2 posts
• Page 1 of 1
having sip response 410 issue
by goose36 » Tue Nov 25, 2008 10:09 am
- goose36
- Posts: 27
- Joined: Thu Oct 18, 2007 3:27 pm
by mcargile » Tue Nov 25, 2008 1:46 pm
This is most likely a sip configuration error on your end, their end, or they have disabled your account.
I would first verify with your Telco that you have your sip settings correct. Most good sip providers have someone who has dealt with Asterisk before and show know what needs to be set to work with their system.
Unfortunately with a bunch of telco's they tend to believe the issue is on your system till you prove them otherwise. If you feel that this is the case try running the following command in the asterisk cli:
This will turn on sip debugging. Then try and place a test call. This will show you all of the control packets you are sending to your carrier. This information is very useful to show the carrier what is going on.
To turn of sip debugging run:
If you are still unable to resolve this with your carrier I would suggest finding a different one.
I would first verify with your Telco that you have your sip settings correct. Most good sip providers have someone who has dealt with Asterisk before and show know what needs to be set to work with their system.
Unfortunately with a bunch of telco's they tend to believe the issue is on your system till you prove them otherwise. If you feel that this is the case try running the following command in the asterisk cli:
- Code: Select all
sip debug
This will turn on sip debugging. Then try and place a test call. This will show you all of the control packets you are sending to your carrier. This information is very useful to show the carrier what is going on.
To turn of sip debugging run:
- Code: Select all
sip no debug
If you are still unable to resolve this with your carrier I would suggest finding a different one.
Michael Cargile | Director of Engineering | ViciDialGroup | http://www.vicidial.com
The official source for VICIDIAL services and support. 1-888-894-VICI (8424)
The official source for VICIDIAL services and support. 1-888-894-VICI (8424)
- mcargile
- Site Admin
- Posts: 615
- Joined: Tue Jan 16, 2007 9:38 am
2 posts
• Page 1 of 1
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