Problem with server and providers

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Problem with server and providers

Postby goose36 » Thu Sep 11, 2008 2:59 pm

Hi,

I´ve been having some serious problems lately with the providers i´ve been using for our minute usage.. to make a long story short.

We used to have a provder where we would call and not have problems at all.. We used to have the dial level at like 5.0 up to 10.0 with 7 agents average daily.. i would increase or decrease the dial level depending on how long it took for my reps to receive a call.. Until at one point the dialer could not complete calls out of the blue and when I checked with putty it would tell me that I would have a busy-circuit issue with the lines congested someting like that.. This happened 3 times already with that provider.. the first two times we did not find any problems because our tech tested the dialer out with a different host and it made the calls.. and when we would switch back to our original provider we would get the busy-circuit again..

The first two times this problem occured we REformated the server reinstalled the vicidial and asterisk.. and we would be working again.. the 3rd time this problem occured our provider said everything was completely fine that it had to be something our tech was doing wrong.. but when our tech checked everything was fine, we tested again with a different provider, and we where able to make calls.. so we decided to switch provder because the other people could not figure anything out at all..



we switched to this provider called Call my way which is a company that took the vicidial source and modified it to their company needs, and they said that my Server willl not hold the amount of agents i have making the amount of calls i want to make and if i try to do make the amount of calls i normally make, chances are i will get a lot of calls broken up, disconnects, or calls getting stuck which is what is happening right now.. but before I NEVER had this problem... just until i switched with them.. since i had 512mb of ram they told me to upgrade to 2gigs which I did.. but htey still said it was not sufficient enough.. and picthed me this duol core dell which had to proccesors of 2.5ghz each..

My server is a Pentium 4 Intel 3ghz, with a HD of 160gbs, and 2gbs of ram.


can someone help me because i dont know what else to do.. my cmpany is losing a lot of money now.. paying techs to try to fix this and nothing being done..


It doesnt make sense how before with the regular vicidial installtion i was able tomake 50 calls, and now with this new service I cant.

Mr.. cargile had said that I must have been throttled by our telco or our telcos telco could be throttling them.. The telco that he was talking about was the one where I had to reformat twice.. Each time I had to reformatt I was being charged 800$.. which means I spent 1600$ in two months trying to fix it.. each time I had to reformatt I would lose 3 days of work just about.. which can cost us our reps becase they need to work to they need to make money..

if someone can provide me some type of explanation or what can be recommended to do..
goose36
 
Posts: 27
Joined: Thu Oct 18, 2007 3:27 pm

Postby mflorell » Thu Sep 11, 2008 4:19 pm

What codec are you using for your SIP trunks?

Are you recording all calls?
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby goose36 » Thu Sep 11, 2008 5:37 pm

g711 i think it is..

and no i am not recording any calls at all.. i dont record because it fill up my server real real quick so i cant record.. the only time i record any call is when the consumer wants the product we record right off the softphone to the agents pc. we use xlite
goose36
 
Posts: 27
Joined: Thu Oct 18, 2007 3:27 pm

Postby mflorell » Fri Sep 12, 2008 5:42 am

A properly configured system should handle that call load with no problem. How many different carriers have you tried?

Where are you calling?
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby goose36 » Mon Sep 15, 2008 8:05 am

I'm calling the US, Whenver I have the Circuit-Busy Congested signal I make an attempt with a different carrier that our tech tests with and its able to make the calls.. then when I switch back to my original carrier that im having a problem with, continues with the same issue. I keep addressing it to my provider and they continue to tell me that on their end everything is perfectly fine, that no changes where made at all, that it has to be on our end.. So then we check back again with our tech and he can't find any problems because we did test it with a different carrier and calls where made. So then he Reformatts the server and when we try to call again, everything is working fine.. After a while it happens again. and then we dont know what its causing it.
goose36
 
Posts: 27
Joined: Thu Oct 18, 2007 3:27 pm

Postby mflorell » Mon Sep 15, 2008 11:54 am

Not sure what else to suggest here, this is something that would require quite a bit of real-time debugging to figure out.
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby goose36 » Mon Sep 15, 2008 7:02 pm

here is the sip debug before i moved to call my way and back to my old provider.. my tech just recently reformatted again our server.. but here is the last sip debug when the problem occured

login as: root
root@172.17.3.85's password:
Last login: Tue Sep 9 09:05:45 2008 from 172.17.3.111

]0;root@vicidialnow:~[root@vicidialnow ~]# asterisk -r

Asterisk 1.2.24, 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 1.2.24 currently running on vicidialnow (pid = 2468)
vicidialnow*CLI>
Verbosity is at least 21

vicidialnow*CLI> sip deubug

vicidialnow*CLI>
SIP Debugging enabled

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/819458032036@default-956f,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/819458032036@default-956f,2", "SIP/eccomm/19458032036|60|tTo") in new stack
We're at 190.10.19.79 port 13054
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:19458032036@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK2a732687;rport

From: "V0909091834000022913" <sip:9178865046@190.10.19.79>;tag=as797c5b03

To: <sip:19458032036@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 184d770713718de724210b306a7954a7@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 13054 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---
-- Called eccomm/19458032036

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/819373429590@default-e718,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/819373429590@default-e718,2", "SIP/eccomm/19373429590|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 11222

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:19373429590@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK3dc8a110;rport

From: "V0909091834000022916" <sip:9178865046@190.10.19.79>;tag=as550c6ffb

To: <sip:19373429590@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 0f34f8510afc50a42e6a83a10e8e4df1@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 11222 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/19373429590

vicidialnow*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=z9hG4bK2a732687;rport=5060

From: "V0909091834000022913" <sip:9178865046@190.10.19.79>;tag=as797c5b03

To: <sip:19458032036@63.251.216.50>

Call-ID: 184d770713718de724210b306a7954a7@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0




--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK2a732687

From: "V0909091834000022913" <sip:9178865046@190.10.19.79>;tag=as797c5b03

To: <sip:19458032036@63.251.216.50>;tag=as4e14b6d7

Contact: <sip:19458032036@63.251.216.23>

Call-ID: 184d770713718de724210b306a7954a7@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:19458032036@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK2a732687;rport

From: "V0909091834000022913" <sip:9178865046@190.10.19.79>;tag=as797c5b03

To: <sip:19458032036@63.251.216.50>;tag=as4e14b6d7

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 184d770713718de724210b306a7954a7@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09d93b70 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/819458032036@default-956f,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 819458032036, 3) exited non-zero on 'Local/819458032036@default-956f,2'
-- Executing DeadAGI("Local/819458032036@default-956f,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/819458032036@default-956f,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>
Destroying call '184d770713718de724210b306a7954a7@190.10.19.79'

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*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=z9hG4bK3dc8a110;rport=5060

From: "V0909091834000022916" <sip:9178865046@190.10.19.79>;tag=as550c6ffb

To: <sip:19373429590@63.251.216.50>

Call-ID: 0f34f8510afc50a42e6a83a10e8e4df1@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0





vicidialnow*CLI>
--- (8 headers 0 lines) ---

vicidialnow*CLI>
-- Executing AGI("Local/817577494406@default-84c8,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK3dc8a110

From: "V0909091834000022916" <sip:9178865046@190.10.19.79>;tag=as550c6ffb

To: <sip:19373429590@63.251.216.50>;tag=as12cbe8ef

Contact: <sip:19373429590@63.251.216.27>

Call-ID: 0f34f8510afc50a42e6a83a10e8e4df1@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER





vicidialnow*CLI>
--- (10 headers 0 lines) ---

vicidialnow*CLI>
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:19373429590@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK3dc8a110;rport

From: "V0909091834000022916" <sip:9178865046@190.10.19.79>;tag=as550c6ffb

To: <sip:19373429590@63.251.216.50>;tag=as12cbe8ef

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 0f34f8510afc50a42e6a83a10e8e4df1@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09da4568 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/819373429590@default-e718,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 819373429590, 3) exited non-zero on 'Local/819373429590@default-e718,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/819373429590@default-e718,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
Destroying call '0f34f8510afc50a42e6a83a10e8e4df1@190.10.19.79'

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/817577494406@default-84c8,2", "SIP/eccomm/17577494406|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 14820

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/819373429590@default-e718,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:17577494406@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK1a46b750;rport

From: "V0909091834000022917" <sip:9178865046@190.10.19.79>;tag=as3f92d4a9

To: <sip:17577494406@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7c15832b7d6e49a2257edcd86be83d32@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 14820 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/17577494406

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/812016504018@default-a2f5,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/812016504018@default-a2f5,2", "SIP/eccomm/12016504018|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 15616

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:12016504018@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK55ff973c;rport

From: "V0909091834000022918" <sip:9178865046@190.10.19.79>;tag=as4fc9d041

To: <sip:12016504018@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7eb9683875b3d4265711fda7740f74f2@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 15616 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/12016504018

vicidialnow*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=z9hG4bK1a46b750;rport=5060

From: "V0909091834000022917" <sip:9178865046@190.10.19.79>;tag=as3f92d4a9

To: <sip:17577494406@63.251.216.50>

Call-ID: 7c15832b7d6e49a2257edcd86be83d32@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0




--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK1a46b750

From: "V0909091834000022917" <sip:9178865046@190.10.19.79>;tag=as3f92d4a9

To: <sip:17577494406@63.251.216.50>;tag=as69137c45

Contact: <sip:17577494406@63.251.216.27>

Call-ID: 7c15832b7d6e49a2257edcd86be83d32@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:17577494406@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK1a46b750;rport

From: "V0909091834000022917" <sip:9178865046@190.10.19.79>;tag=as3f92d4a9

To: <sip:17577494406@63.251.216.50>;tag=as69137c45

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7c15832b7d6e49a2257edcd86be83d32@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09d93b70 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/817577494406@default-84c8,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 817577494406, 3) exited non-zero on 'Local/817577494406@default-84c8,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/817577494406@default-84c8,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/817577494406@default-84c8,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>
Destroying call '7c15832b7d6e49a2257edcd86be83d32@190.10.19.79'

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/817406492973@default-fc64,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/817406492973@default-fc64,2", "SIP/eccomm/17406492973|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 18222

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:17406492973@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK0b2b7dc9;rport

From: "V0909091834000022919" <sip:9178865046@190.10.19.79>;tag=as2519ec6e

To: <sip:17406492973@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 4bcb319e3dc5e67f329462194ae1d921@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 18222 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/17406492973

vicidialnow*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=z9hG4bK55ff973c;rport=5060

From: "V0909091834000022918" <sip:9178865046@190.10.19.79>;tag=as4fc9d041

To: <sip:12016504018@63.251.216.50>

Call-ID: 7eb9683875b3d4265711fda7740f74f2@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0





vicidialnow*CLI>
--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK55ff973c

From: "V0909091834000022918" <sip:9178865046@190.10.19.79>;tag=as4fc9d041

To: <sip:12016504018@63.251.216.50>;tag=as1ac26120

Contact: <sip:12016504018@63.251.216.27>

Call-ID: 7eb9683875b3d4265711fda7740f74f2@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER





vicidialnow*CLI>
--- (10 headers 0 lines) ---

vicidialnow*CLI>
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:12016504018@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK55ff973c;rport

From: "V0909091834000022918" <sip:9178865046@190.10.19.79>;tag=as4fc9d041

To: <sip:12016504018@63.251.216.50>;tag=as1ac26120

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7eb9683875b3d4265711fda7740f74f2@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09da4568 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/812016504018@default-a2f5,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 812016504018, 3) exited non-zero on 'Local/812016504018@default-a2f5,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/812016504018@default-a2f5,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/812016504018@default-a2f5,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/813369453080@default-7238,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/813369453080@default-7238,2", "SIP/eccomm/13369453080|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 12042

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:13369453080@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK48bb5f18;rport

From: "V0909091834000022922" <sip:9178865046@190.10.19.79>;tag=as166ba1a9

To: <sip:13369453080@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7a7c52287392a7f30a2550d2195a4923@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 12042 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/13369453080

vicidialnow*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=z9hG4bK0b2b7dc9;rport=5060

From: "V0909091834000022919" <sip:9178865046@190.10.19.79>;tag=as2519ec6e

To: <sip:17406492973@63.251.216.50>

Call-ID: 4bcb319e3dc5e67f329462194ae1d921@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0




--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK0b2b7dc9

From: "V0909091834000022919" <sip:9178865046@190.10.19.79>;tag=as2519ec6e

To: <sip:17406492973@63.251.216.50>;tag=as285d36ed

Contact: <sip:17406492973@63.251.216.25>

Call-ID: 4bcb319e3dc5e67f329462194ae1d921@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:17406492973@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK0b2b7dc9;rport

From: "V0909091834000022919" <sip:9178865046@190.10.19.79>;tag=as2519ec6e

To: <sip:17406492973@63.251.216.50>;tag=as285d36ed

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 4bcb319e3dc5e67f329462194ae1d921@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---
Destroying call '7eb9683875b3d4265711fda7740f74f2@190.10.19.79'

vicidialnow*CLI>
-- SIP/eccomm-09d93b70 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/817406492973@default-fc64,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 817406492973, 3) exited non-zero on 'Local/817406492973@default-fc64,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/817406492973@default-fc64,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/817406492973@default-fc64,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*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=z9hG4bK48bb5f18;rport=5060

From: "V0909091834000022922" <sip:9178865046@190.10.19.79>;tag=as166ba1a9

To: <sip:13369453080@63.251.216.50>

Call-ID: 7a7c52287392a7f30a2550d2195a4923@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0




--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK48bb5f18

From: "V0909091834000022922" <sip:9178865046@190.10.19.79>;tag=as166ba1a9

To: <sip:13369453080@63.251.216.50>;tag=as631f4d36

Contact: <sip:13369453080@63.251.216.25>

Call-ID: 7a7c52287392a7f30a2550d2195a4923@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:13369453080@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK48bb5f18;rport

From: "V0909091834000022922" <sip:9178865046@190.10.19.79>;tag=as166ba1a9

To: <sip:13369453080@63.251.216.50>;tag=as631f4d36

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7a7c52287392a7f30a2550d2195a4923@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09da9aa8 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/813369453080@default-7238,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 813369453080, 3) exited non-zero on 'Local/813369453080@default-7238,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/813369453080@default-7238,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/813369453080@default-7238,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>
Destroying call '7a7c52287392a7f30a2550d2195a4923@190.10.19.79'

vicidialnow*CLI>
Destroying call '4bcb319e3dc5e67f329462194ae1d921@190.10.19.79'

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/813362093781@default-1d69,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/813362093781@default-1d69,2", "SIP/eccomm/13362093781|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 17960

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:13362093781@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK3ddd3663;rport

From: "V0909091838000022924" <sip:9178865046@190.10.19.79>;tag=as636f90a8

To: <sip:13362093781@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7463c236543dc502630e906b5f241396@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:38 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 17960 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/13362093781

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/816092339276@default-3a11,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*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=z9hG4bK3ddd3663;rport=5060

From: "V0909091838000022924" <sip:9178865046@190.10.19.79>;tag=as636f90a8

To: <sip:13362093781@63.251.216.50>

Call-ID: 7463c236543dc502630e906b5f241396@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0




--- (8 headers 0 lines) ---

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/816092339276@default-3a11,2", "SIP/eccomm/16092339276|60|tTo") in new stack

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK3ddd3663

From: "V0909091838000022924" <sip:9178865046@190.10.19.79>;tag=as636f90a8

To: <sip:13362093781@63.251.216.50>;tag=as2c8b851b

Contact: <sip:13362093781@63.251.216.27>

Call-ID: 7463c236543dc502630e906b5f241396@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:13362093781@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK3ddd3663;rport

From: "V0909091838000022924" <sip:9178865046@190.10.19.79>;tag=as636f90a8

To: <sip:13362093781@63.251.216.50>;tag=as2c8b851b

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7463c236543dc502630e906b5f241396@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09d93b70 is circuit-busy

vicidialnow*CLI>
We're at 190.10.19.79 port 16720

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:16092339276@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK251579f9;rport

From: "V0909091838000022925" <sip:9178865046@190.10.19.79>;tag=as08738de1

To: <sip:16092339276@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 275e43d8191e870c547ed25b53383a93@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:38 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 16720 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/16092339276

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/813362093781@default-1d69,2", "") in new stack
== Spawn extension (default, 813362093781, 3) exited non-zero on 'Local/813362093781@default-1d69,2'
-- Executing DeadAGI("Local/813362093781@default-1d69,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/813362093781@default-1d69,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>
Destroying call '7463c236543dc502630e906b5f241396@190.10.19.79'

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/815704991557@default-0c61,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/815704991557@default-0c61,2", "SIP/eccomm/15704991557|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 19866

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:15704991557@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK54501929;rport

From: "V0909091838000022927" <sip:9178865046@190.10.19.79>;tag=as2af845e8

To: <sip:15704991557@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7c8d69962c9a94a536cc805d0297b1fb@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:38 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 19866 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/15704991557

vicidialnow*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=z9hG4bK251579f9;rport=5060

From: "V0909091838000022925" <sip:9178865046@190.10.19.79>;tag=as08738de1

To: <sip:16092339276@63.251.216.50>

Call-ID: 275e43d8191e870c547ed25b53383a93@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0




--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK251579f9

From: "V0909091838000022925" <sip:9178865046@190.10.19.79>;tag=as08738de1

To: <sip:16092339276@63.251.216.50>;tag=as516393c9

Contact: <sip:16092339276@63.251.216.25>

Call-ID: 275e43d8191e870c547ed25b53383a93@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:16092339276@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK251579f9;rport

From: "V0909091838000022925" <sip:9178865046@190.10.19.79>;tag=as08738de1

To: <sip:16092339276@63.251.216.50>;tag=as516393c9

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 275e43d8191e870c547ed25b53383a93@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09da4568 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/816092339276@default-3a11,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 816092339276, 3) exited non-zero on 'Local/816092339276@default-3a11,2'
-- Executing DeadAGI("Local/816092339276@default-3a11,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/816092339276@default-3a11,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
Destroying call '275e43d8191e870c547ed25b53383a93@190.10.19.79'

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*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=z9hG4bK54501929;rport=5060

From: "V0909091838000022927" <sip:9178865046@190.10.19.79>;tag=as2af845e8

To: <sip:15704991557@63.251.216.50>

Call-ID: 7c8d69962c9a94a536cc805d0297b1fb@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0





vicidialnow*CLI>
--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK54501929

From: "V0909091838000022927" <sip:9178865046@190.10.19.79>;tag=as2af845e8

To: <sip:15704991557@63.251.216.50>;tag=as56ca01e6

Contact: <sip:15704991557@63.251.216.23>

Call-ID: 7c8d69962c9a94a536cc805d0297b1fb@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:15704991557@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK54501929;rport

From: "V0909091838000022927" <sip:9178865046@190.10.19.79>;tag=as2af845e8

To: <sip:15704991557@63.251.216.50>;tag=as56ca01e6

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 7c8d69962c9a94a536cc805d0297b1fb@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09d93b70 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/815704991557@default-0c61,2", "") in new stack
== Spawn extension (default, 815704991557, 3) exited non-zero on 'Local/815704991557@default-0c61,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/815704991557@default-0c61,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/815704991557@default-0c61,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
Destroying call '7c8d69962c9a94a536cc805d0297b1fb@190.10.19.79'

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>

<-- SIP read from 172.17.3.111:41364:





--- (0 headers 1 lines) ---

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/819043336545@default-fa93,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/819043336545@default-fa93,2", "SIP/eccomm/19043336545|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 19948

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:19043336545@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5c225bf7;rport

From: "V0909091840000022929" <sip:9178865046@190.10.19.79>;tag=as1c273abc

To: <sip:19043336545@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 2b3c7341173802b273e28e276e5ee04f@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:41 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 19948 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/19043336545

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/816107198484@default-d915,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*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=z9hG4bK5c225bf7;rport=5060

From: "V0909091840000022929" <sip:9178865046@190.10.19.79>;tag=as1c273abc

To: <sip:19043336545@63.251.216.50>

Call-ID: 2b3c7341173802b273e28e276e5ee04f@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0




--- (8 headers 0 lines) ---

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5c225bf7

From: "V0909091840000022929" <sip:9178865046@190.10.19.79>;tag=as1c273abc

To: <sip:19043336545@63.251.216.50>;tag=as42360c6e

Contact: <sip:19043336545@63.251.216.23>

Call-ID: 2b3c7341173802b273e28e276e5ee04f@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:19043336545@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK5c225bf7;rport

From: "V0909091840000022929" <sip:9178865046@190.10.19.79>;tag=as1c273abc

To: <sip:19043336545@63.251.216.50>;tag=as42360c6e

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 2b3c7341173802b273e28e276e5ee04f@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09d93b70 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/819043336545@default-fa93,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 819043336545, 3) exited non-zero on 'Local/819043336545@default-fa93,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/819043336545@default-fa93,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/816107198484@default-d915,2", "SIP/eccomm/16107198484|60|tTo") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/819043336545@default-fa93,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 16730

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:16107198484@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK293a3f74;rport

From: "V0909091840000022930" <sip:9178865046@190.10.19.79>;tag=as29c7bbff

To: <sip:16107198484@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 6ef5a4127bd955b763a99613580d5849@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:41 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 16730 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/16107198484

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>
Destroying call '2b3c7341173802b273e28e276e5ee04f@190.10.19.79'

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/813307948486@default-adb7,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/813307948486@default-adb7,2", "SIP/eccomm/13307948486|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 13580

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:13307948486@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK155cad22;rport

From: "V0909091840000022933" <sip:9178865046@190.10.19.79>;tag=as4b101e87

To: <sip:13307948486@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 021859a7569e3a513aa4a1876202e55f@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:41 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 13580 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/13307948486

vicidialnow*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=z9hG4bK293a3f74;rport=5060

From: "V0909091840000022930" <sip:9178865046@190.10.19.79>;tag=as29c7bbff

To: <sip:16107198484@63.251.216.50>

Call-ID: 6ef5a4127bd955b763a99613580d5849@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0





vicidialnow*CLI>
--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK293a3f74

From: "V0909091840000022930" <sip:9178865046@190.10.19.79>;tag=as29c7bbff

To: <sip:16107198484@63.251.216.50>;tag=as4616ae5e

Contact: <sip:16107198484@63.251.216.23>

Call-ID: 6ef5a4127bd955b763a99613580d5849@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER





vicidialnow*CLI>
--- (10 headers 0 lines) ---

vicidialnow*CLI>
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:16107198484@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK293a3f74;rport

From: "V0909091840000022930" <sip:9178865046@190.10.19.79>;tag=as29c7bbff

To: <sip:16107198484@63.251.216.50>;tag=as4616ae5e

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 6ef5a4127bd955b763a99613580d5849@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-b7a00510 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/816107198484@default-d915,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 816107198484, 3) exited non-zero on 'Local/816107198484@default-d915,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/816107198484@default-d915,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
Destroying call '6ef5a4127bd955b763a99613580d5849@190.10.19.79'

vicidialnow*CLI>
-- Executing DeadAGI("Local/816107198484@default-d915,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*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=z9hG4bK155cad22;rport=5060

From: "V0909091840000022933" <sip:9178865046@190.10.19.79>;tag=as4b101e87

To: <sip:13307948486@63.251.216.50>

Call-ID: 021859a7569e3a513aa4a1876202e55f@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0




--- (8 headers 0 lines) ---

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK155cad22

From: "V0909091840000022933" <sip:9178865046@190.10.19.79>;tag=as4b101e87

To: <sip:13307948486@63.251.216.50>;tag=as33a6a38d

Contact: <sip:13307948486@63.251.216.25>

Call-ID: 021859a7569e3a513aa4a1876202e55f@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:13307948486@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK155cad22;rport

From: "V0909091840000022933" <sip:9178865046@190.10.19.79>;tag=as4b101e87

To: <sip:13307948486@63.251.216.50>;tag=as33a6a38d

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 021859a7569e3a513aa4a1876202e55f@190.10.19.79

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---

vicidialnow*CLI>
-- SIP/eccomm-09d93b70 is circuit-busy

vicidialnow*CLI>
== Everyone is busy/congested at this time (1:0/1/0)

vicidialnow*CLI>
-- Executing Hangup("Local/813307948486@default-adb7,2", "") in new stack

vicidialnow*CLI>
== Spawn extension (default, 813307948486, 3) exited non-zero on 'Local/813307948486@default-adb7,2'

vicidialnow*CLI>
-- Executing DeadAGI("Local/813307948486@default-adb7,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing DeadAGI("Local/813307948486@default-adb7,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

vicidialnow*CLI>
Destroying call '021859a7569e3a513aa4a1876202e55f@190.10.19.79'

vicidialnow*CLI>
== Manager 'sendcron' logged off from 127.0.0.1

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/816105821256@default-3ec7,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

vicidialnow*CLI>
-- Executing Dial("Local/816105821256@default-3ec7,2", "SIP/eccomm/16105821256|60|tTo") in new stack

vicidialnow*CLI>
We're at 190.10.19.79 port 14656

vicidialnow*CLI>
Adding codec 0x4 (ulaw) to SDP

vicidialnow*CLI>
Adding codec 0x100 (g729) to SDP

vicidialnow*CLI>
Adding codec 0x2 (gsm) to SDP

vicidialnow*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

vicidialnow*CLI>
13 headers, 13 lines

vicidialnow*CLI>
Reliably Transmitting (no NAT) to 63.251.216.50:5060:
INVITE sip:16105821256@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK0114316c;rport

From: "V0909091843000022935" <sip:9178865046@190.10.19.79>;tag=as6bc48814

To: <sip:16105821256@63.251.216.50>

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 0c7359626d447c167aa449fb7731b391@190.10.19.79

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 09 Sep 2008 13:18:43 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 284



v=0

o=root 2468 2468 IN IP4 190.10.19.79

s=session

c=IN IP4 190.10.19.79

t=0 0

m=audio 14656 RTP/AVP 0 18 3 101

a=rtpmap:0 PCMU/8000

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 - - - -


---

vicidialnow*CLI>
-- Called eccomm/16105821256

vicidialnow*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=z9hG4bK0114316c;rport=5060

From: "V0909091843000022935" <sip:9178865046@190.10.19.79>;tag=as6bc48814

To: <sip:16105821256@63.251.216.50>

Call-ID: 0c7359626d447c167aa449fb7731b391@190.10.19.79

CSeq: 102 INVITE

Server: XCast Partner/1.1

Content-Length: 0





vicidialnow*CLI>
--- (8 headers 0 lines) ---

vicidialnow*CLI>

== Parsing '/etc/asterisk/manager.conf': Found

vicidialnow*CLI>
== Manager 'sendcron' logged on from 127.0.0.1

vicidialnow*CLI>
-- Executing AGI("Local/814196090347@default-c893,2", "agi://127.0.0.1:4577/call_log") in new stack

vicidialnow*CLI>

<-- SIP read from 63.251.216.50:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK0114316c

From: "V0909091843000022935" <sip:9178865046@190.10.19.79>;tag=as6bc48814

To: <sip:16105821256@63.251.216.50>;tag=as30b7af07

Contact: <sip:16105821256@63.251.216.27>

Call-ID: 0c7359626d447c167aa449fb7731b391@190.10.19.79

CSeq: 102 INVITE

User-Agent: SIPTalk Broker

Content-Length: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER




--- (10 headers 0 lines) ---

vicidialnow*CLI>
Transmitting (no NAT) to 63.251.216.50:5060:
ACK sip:16105821256@63.251.216.50 SIP/2.0

Via: SIP/2.0/UDP 190.10.19.79:5060;branch=z9hG4bK0114316c;rport

From: "V0909091843000022935" <sip:9178865046@190.10.19.79>;tag=as6bc48814

To: <sip:16105821256@63.251.216.50>;tag=as30b7af07

Contact: <sip:9178865046@190.10.19.79>

Call-ID: 0c7359626d447c167aa449fb7731b39
goose36
 
Posts: 27
Joined: Thu Oct 18, 2007 3:27 pm

Postby mflorell » Tue Sep 16, 2008 3:23 am

Not really sure why you posted hundreds of lines of SIP debug output here. As I had mentioned this would really require real-time debugging to figure out what's going on.
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby goose36 » Tue Sep 16, 2008 11:26 am

ohhhhh.. ijust though that would be helpful..
if the problem ever occurs again, what can i do for a real time debugg.. i dont have a tech and each time i hire a tech, here in costa rica they shove it up your ass becuase your a US citizen and what not and i always end up paying tons of money for each problem that occurs. i mean
goose36
 
Posts: 27
Joined: Thu Oct 18, 2007 3:27 pm

Postby mflorell » Tue Sep 16, 2008 2:02 pm

There are a lot of things to check, but first get specific phone numbers and check the Asterisk debug logs for the entire call history of those calls, then depending on what you find there, you need to check several other things.

As for cost of our services(as the VICIDIAL Group), we don't care your nationality or place of residence, we charge everybody the same $200/hour for our services, we do offer volume discounts though.
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby goose36 » Wed Sep 17, 2008 11:07 am

how can I contact you or anyone from the vicidial group, whenever this problem occurs again where i can´t fix the issue..
goose36
 
Posts: 27
Joined: Thu Oct 18, 2007 3:27 pm

Postby mflorell » Wed Sep 17, 2008 11:39 am

you can go to our website:
http://www.vicidial.com

We have a contact form and several phone numbers
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida


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