Changing Carrier in vicidial

[Cj4life24]

I am about to change carrier to a new provider for my VOIP service and about to make the change to inbound calling as well. I need a lillte direction. My new provider is askin what our number of DINT and what protocol we use int or int 2 where can i find the information to give to thenm so they can finish setting up out trunk so i can make the change.

My next question is in my carrier set up can i just copy the dialplan entry box over to my carrier set up dialplan box? can some one pleas give me a little direction

[williamconley]

I'm not sure how you managed OpenSuSE 11.2 when Vicibox Redux doesn't install that. But that's beside the point. LOL

To create a new carrier, you would copy it but you would do best to copy it in a way that allows both to exist at the same time (this allows an unlimited number of carriers on each machine, simultaneously). It is possible to use different carriers on the same campaign (by having the dialplan send through different carriers on different servers even though the agents on those dialers are on the same campaign!). In a more normal single-server system, it is entirely possible in a regular setup to have multiple campaigns each with its own Carrier (so you have three campaigns running in three different carriers all at the same time on the same dialer).

To be sure this works, each Carrier should have a different "Dial Prefix". Each "Dialplan" would begin with a different number, which represents the Dial Prefix you would put in the Campaign to use that Carrier.

Then changing between carriers (whether for the day, or just for a test, or because you are changing over permanently if it "works") is as simple as changing a single digit in the Dial Prefix of each campaign. And if it FAILS, you can change it back a few seconds later and your agents may never even know it happened (business as usual).

If your dialplan is like this (this pattern appears on each line f the dialplan):

Code: Select all

_91NXXNXXXXXX

You'll add a new one with this instead:

Code: Select all

_81NXXNXXXXXX

Now instead of "9" for the dial prefix, the new one will use "8".

But: Be sure the "dial" line of the dialplan is unique as well. Do not use the same "Global Variable" in the new one, or it will try to dial through the previous carrier. So:

Code: Select all

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${DIAL9TRUNK}/${EXTEN:1},55,tTor)
exten => _91NXXNXXXXXX,3,Hangup

May become:

Code: Select all

exten => _81NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81NXXNXXXXXX,2,Dial(${DIAL8TRUNK}/${EXTEN:1},55,tTor)
exten => _81NXXNXXXXXX,3,Hangup

Oh: And I've never heard of "int1 or int2"? Who is this carrier that would ask such a question?

[Cj4life24]

Ok i really appriciate the help. INT or INT2 is a protocol for the carrire thought we used for the PRI. And i know i run REDUX 3.1.9 and you always post about my open suse. The reason i think i am running it i think REDux is installed on top of it because when i reboot the Vicidial Server when it starts to come back up the first screen displayed after the splash screen is and Open Suse 11.2 Boot menue with only 2 options Failsafe Open Suse 11.2 and the option 2 is just Open SUSE 11.2 the default boot is into the open suse 11.2 option 2 but when it boots after selection that option it boots into Redux 3.1.9 so that is why i feel like i am running it (boot menu is Very similar to a Dual Boot Windows Linux Machine. ANd it sucks inheriting a mess from the previous Admin and never used Vicidial before but very easy to learn. So any ways

I got my registration string and everything form my carrier And I did what you posted but got an error Cause 21 Congestion call rejected. Now i do want to tell you that currently both carriers are over the same physical T-1 trunk and i know it can only handle 24 consecutive calls and the and i tried doing it during a shift were calls were being made. My question is "is it just because i am overloading the current T-1 trunk so i can't complete a dial . Is it probably best to test off hours?

[Cj4life24]

ok Update:

tested quickly while agents were at lunch got same error but noticed that the the two sounds heard by agents one when call is connected and one when call is disconnected are heard back to back like the calls is connected but then disconnects instantly. So i am pretty sure the call is just not making it out on the trunk is that correct in thinking the call is terminated before it hits the trunk?

New dial plan i created as per your instructions

exten=>_81NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_81NXXNXXXXXX,2,Dial(${XXXXXXX}/${EXTEN:2},,tTor)
exten=>_81NXXNXXXXXX,3,Hangup

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

Dial Plan I coppied and changed

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

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


Now My account entry that i coppied from Current carrier. This will be pretty much identical except for the Username, Secret, Host and Fromdomain lines is that correct? all my new carrier gave me was Registration string, host, username,and secrete. is that all i need.

[My host]
type=peer
username=UN
secret=PW
host=myhost.com
dtmfmode=rfc2833
context=truckinbound
insecure=port,invite
usercallerid=yes
trustrpid=yes
sendrpid=no
disallow=all
allow=ulaw
fromdomain=myhost.com

Any Sugestions or help would be greatly appriciated

[DomeDan]

install ngrep-sip: "zypper install ngrep-sip"
and run it when your testing.

change or remove these lines:
host=myhost.com
fromdomain=myhost.com

do you need this?:
usercallerid=yes
trustrpid=yes
sendrpid=no

use google to find out what every setting means

[williamconley]

T1 calls are not the same as SIP or IAX calls and are handled by a different method. Which really makes me wonder why you mentioned a VOIP carrier earlier ... unless you are using the T1 for "net" and not for "voice".

You will need to supply an asterisk command line execution sample of an attempted call (not 3000 lines of unrelated code from other active calls, just the CLI from one attempt/fail).

[Cj4life24]

Ok i think this is the code you need and all of it not a bunch of other unrelated code.


[Oct 24 08:36:07] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 24 08:36:07] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-c978,2", "8600051|F") in new stack
[Oct 24 08:36:07] > Channel Local/8600051@default-c978,1 was answered.
[Oct 24 08:36:07] -- Executing [818134746916@default:1] AGI("Local/8600051@default-c978,1", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 24 08:36:08] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 24 08:36:08] -- Executing [818134746916@default:2] Dial("Local/8600051@default-c978,1", "SIP/telovations/8134746916||tTor") in new stack
[Oct 24 08:36:08] -- Called telovations/8134746916
[Oct 24 08:36:08] WARNING[2950]: chan_sip.c:13482 handle_response_invite: Received response: "Forbidden" from '"M0240836060000357469" <sip:3037315977@71.101.88.206>;tag=as726c49d7'
[Oct 24 08:36:08] -- SIP/telovations-0001d6fd is circuit-busy
[Oct 24 08:36:08] == Everyone is busy/congested at this time (1:0/1/0)
[Oct 24 08:36:08] -- Executing [818134746916@default:3] Hangup("Local/8600051@default-c978,1", "") in new stack
[Oct 24 08:36:08] == Spawn extension (default, 818134746916, 3) exited non-zero on 'Local/8600051@default-c978,1'
[Oct 24 08:36:08] -- Executing [h@default:1] DeadAGI("Local/8600051@default-c978,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CO NGESTION----------") in new stack
[Oct 24 08:36:08] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 24 08:36:08] == Spawn extension (default, 8600051, 1) exited non-zero on'Local/8600051@default-c978,2'
[Oct 24 08:36:08] -- Executing [h@default:1] DeadAGI("Local/8600051@default-c978,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-------- -------") in new stack

I hope this is what you were looking for and i hope i can get this solved now to move forward and start using my new carrier. Thanks so much for the help

[williamconley]

telovations says Forbidden.

It is not possible for you to "override" that from your end (you cannot force them to complete the call ...). You have to ask telovations WHY they are rejecting your call.

You can also try using "sip set debug on" or "sip set debug" depending on your asterisk version to (perhaps) get an "english" version of the code before Forbidden is extracted ... but it is entirely likely that it will be simply "forbidden" in the sip debug as well.

In which case you have to ask them why.

They may be refusing the call for any number of reasons (your user name is wrong, you did not pay your phone bill, they did not complete your carrier setup yet, you are sending to the wrong server, you are missing a sip.conf configuration setting that they require, or have one they do not allow ...). No way to tell without asking them.

[Cj4life24]

So i talked to my carrier and he got me the proper resistration string.

here is the code for anothe call still doing same thing "Call Rejected: Congestion cause 21 call rejected"

[Oct 25 10:01:06] --- (10 headers 0 lines) ---
[Oct 25 10:01:06] Really destroying SIP dialog '6b4e41795030c4ef196ddb5a6eaf7bf8@192.168.1.2' Method: OPTIONS
[Oct 25 10:01:07] == Parsing '/etc/asterisk/manager.conf': [Oct 25 10:01:07] Found
[Oct 25 10:01:07] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 25 10:01:07] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-72c8,2", "8600051|Fmq") in new stack
[Oct 25 10:01:07] > Channel Local/58600051@default-72c8,1 was answered.
[Oct 25 10:01:07] -- Executing [8309@default:1] Answer("Local/58600051@default-72c8,1", "") in new stack
[Oct 25 10:01:07] -- Executing [8309@default:2] Monitor("Local/58600051@default-72c8,1", "wav|20121025-100106_8134746916_444") in new stack
[Oct 25 10:01:07] -- Executing [8309@default:3] Wait("Local/58600051@default-72c8,1", "3600") in new stack
[Oct 25 10:01:07] Reliably Transmitting (NAT) to 199.255.232.70:5060:
OPTIONS sip:voip.telovations.net;cpd=on SIP/2.0
Via: SIP/2.0/UDP 71.101.88.206:5060;branch=z9hG4bK1817dc15;rport
From: "asterisk" <sip:asterisk@71.101.88.206>;tag=as3ee33cd0
To: <sip:voip.telovations.net;cpd=on>
Contact: <sip:asterisk@71.101.88.206>
Call-ID: 17cc03e27d5ec2f25e02bc016a1cd66c@71.101.88.206
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Oct 2012 14:01:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

[Cj4life24]

Got it registered it was a problem on their end the registration string was incorrect. i added the secret to the string and got it registerd. But now when i tried to make a call i get this error.

"Call Rejected:Chanunavail
Cause 1-Unallocated(unassigned) Number

and a recording says not a valid extention

here is the code for a call


[Oct 25 12:05:04] --- (10 headers 0 lines) ---
[Oct 25 12:05:04] Really destroying SIP dialog '00bc1985294303363f8e374e24ec6c1f@192.168.1.2' Method: OPTIONS
[Oct 25 12:05:04] Really destroying SIP dialog '2c53afd3255816834a79b3af2536ad30@192.168.1.2' Method: OPTIONS
[Oct 25 12:05:05] == Parsing '/etc/asterisk/manager.conf': [Oct 25 12:05:05] Found
[Oct 25 12:05:05] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 25 12:05:05] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-9780,2", "8600051|F") in new stack
[Oct 25 12:05:05] > Channel Local/8600051@default-9780,1 was answered.
[Oct 25 12:05:05] == Starting Local/8600051@default-9780,1 at default,818134746916,1 failed so falling back to exten 's'
[Oct 25 12:05:05] == Starting Local/8600051@default-9780,1 at default,s,1 still failed so falling back to context 'default'
[Oct 25 12:05:05] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-9780,1
[Oct 25 12:05:05] -- Executing [i@default:1] Playback("Local/8600051@default-9780,1", "invalid") in new stack
[Oct 25 12:05:05] -- <Local/8600051@default-9780,1> Playing 'invalid' (language 'en')
[Oct 25 12:05:05] WARNING[15955]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Oct 25 12:05:05] WARNING[15955]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Oct 25 12:05:06] == Parsing '/etc/asterisk/manager.conf': [Oct 25 12:05:06] Found
[Oct 25 12:05:06] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 25 12:05:06] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 25 12:05:07] Reliably Transmitting (NAT) to 192.168.1.59:5060:
OPTIONS sip:235@192.168.1.59;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48197b93;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as3259ab89
To: <sip:235@192.168.1.59;cpd=on>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 71e255441231467b52f7acc81914c0ad@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Oct 2012 16:05:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

[williamconley]

show your dialplan for this carrier.

if it does not start with "81" ... that's why it's failing ... you are dialing 8+1+10digits ... so your dialplan would need to have 81NXXNXXXXXX on all three lines to "match up" with what you are dialing.

chanunavailable is "i have no path to dial this number" (no matching extension pattern).

[Cj4life24]

Dial Plan was set at 8 i poseted it in an above post i followed your directions on that one so i could try and have the 2 carriers simoutanisoly. This morning i tried changing it to 9 and tried a call and got same error. here is the dial plan as it is currently and also so CLI code for a call useing the current dial plan.



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

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





[Oct 26 09:25:39] == Refreshing DNS lookups.
[Oct 26 09:25:48] == Parsing '/etc/asterisk/manager.conf':
[Oct 26 09:25:48] Found
[Oct 26 09:25:48] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 09:25:50] > Channel SIP/219-0000057c was answered.
[Oct 26 09:25:50] -- Executing [8600051@default:1] MeetMe("SIP/219-0000057c", "8600051|F") in new stack
[Oct 26 09:25:50] == Parsing '/etc/asterisk/meetme.conf': [Oct 26 09:25:50] Found
[Oct 26 09:25:50] == Parsing '/etc/asterisk/meetme-vicidial.conf':
[Oct 26 09:25:50] Found
[Oct 26 09:25:50] -- Created MeetMe conference 1023 for conference '8600051'
[Oct 26 09:25:50] -- <SIP/219-0000057c> Playing 'conf-onlyperson' (language'en')
[Oct 26 09:25:52] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 09:26:01] == Parsing '/etc/asterisk/manager.conf':
[Oct 26 09:26:01] Found
[Oct 26 09:26:01] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 09:26:01] == Parsing '/etc/asterisk/manager.conf':
[Oct 26 09:26:01] Found
[Oct 26 09:26:01] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 09:26:01] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 09:26:03] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 09:26:06] == Parsing '/etc/asterisk/manager.conf':
[Oct 26 09:26:06] Found
[Oct 26 09:26:06] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 09:26:06] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 09:26:19] == Parsing '/etc/asterisk/manager.conf':
[Oct 26 09:26:19] Found
[Oct 26 09:26:19] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 09:26:19] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-80ed,2", "8600051|F") in new stack
[Oct 26 09:26:19] > Channel Local/8600051@default-80ed,1 was answered.
[Oct 26 09:26:19] -- Executing [918134746916@default:1] AGI("Local/8600051@default-80ed,1", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 26 09:26:20] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 26 09:26:20] -- Executing [918134746916@default:2] Dial("Local/8600051@default-80ed,1", "SIP/telovations/8134746916||tTor") in new stack
[Oct 26 09:26:20] -- Called telovations/8134746916
[Oct 26 09:26:20] -- Got SIP response 604 "Does Not Exist Anywhere" back from 208.82.176.73
[Oct 26 09:26:20] == Everyone is busy/congested at this time (1:0/0/1)
[Oct 26 09:26:20] -- Executing [918134746916@default:3] Hangup("Local/8600051@default-80ed,1", "") in new stack
[Oct 26 09:26:20] == Spawn extension (default, 918134746916, 3) exited non-zero on 'Local/8600051@default-80ed,1'
[Oct 26 09:26:20] -- Executing [h@default:1] DeadAGI("Local/8600051@default-80ed,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL----------") in new stack
[Oct 26 09:26:20] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----1-----CHANUNAVAIL---------- completed, returning 0
[Oct 26 09:26:20] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-80ed,2'
[Oct 26 09:26:20] -- Executing [h@default:1] DeadAGI("Local/8600051@default-80ed,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct 26 09:26:20] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----0--------------- completed, returning 0
[Oct 26 09:26:21] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 09:26:25] == Parsing '/etc/asterisk/manager.conf'
[Oct 26 09:26:25] Found
[Oct 26 09:26:25] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 09:26:25] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-e1dd,2", "8600051|Fmq") in new stack
[Oct 26 09:26:25] > Channel Local/58600051@default-e1dd,1 was answered.
[Oct 26 09:26:25] -- Executing [8309@default:1] Answer("Local/58600051@default-e1dd,1", "") in new stack
[Oct 26 09:26:25] -- Executing [8309@default:2] Monitor("Local/58600051@default-e1dd,1", "wav|20121026-092624_8134746916_444") in new stack
[Oct 26 09:26:25] -- Executing [8309@default:3] Wait("Local/58600051@default-e1dd,1", "3600") in new stack
[Oct 26 09:26:27] == Manager 'sendcron' logged off from 127.0.0.1


I really appriciate your help man i will check back with the carrier today and see if i can find out y i don't have a path out.

[williamconley]

[Oct 26 09:26:20] -- Executing [918134746916@default:2] Dial("Local/8600051@default-80ed,1", "SIP/telovations/8134746916||tTor") in new stack
[Oct 26 09:26:20] -- Called telovations/8134746916
[Oct 26 09:26:20] -- Got SIP response 604 "Does Not Exist Anywhere" back from 208.82.176.73

Carrier says they don't know how to dial this number.

[Cj4life24]

Man i really appriciate your help. I got it all figured out atleast for the outbound calls. I guess i was not sending the provider in caller ID a number that was registered to the trunk. I am able to now make outbound calls. Now for the Fun part trying to take this trunk and use it for inbound calls and being able to transfer call through the server rather than taking inbound calls on analog lines then calling them back. Thanks again man for taking the time to help me out.

[williamconley]

this is not a pstn line. there really are no "trunks" per se.

your ability to use "this" trunk for inbound now that it works for outbound is related only to two things: 1) convince your telco (voip provider) to send the call to this server and then 2) convince you asterisk server to send the call to the Vicidial scripts for control.

1) is handled either by modifying a configuration at your voip provider (tech support, web interface ... however they set it up) to convince the call to go to the IP address of your server OR (in many cases) you can "Register" to an account at the provider which is a way of saying "send my calls here". In that case, the phone number would be assigned to an account, the account would have a user/pass and when you "register", the voip provider automatically sends calls to the ip address you registered from (your server basically opens a hole in your firewall that your voip provider sends calls back through as long as your registration remains active).

2) once you have convinced the provider to send the calls to your machine, they will arrive at your IP address. if you are registering, that process will open a hole in the firewall. IF not registered, your server will need access to port 5060 (which is dangerous to leave open but that's a different story, get a good firewall installed in your system!). Once you have port 5060 available to your vicidial server, the call will arrive in asterisk and asterisk will check "sip.conf" and "sip-vicidial.conf" for a matching "host" entry. When it finds an IP address that matches the sender of the call, it will check the "context=" entry and send the call to that context. All sip carrier accounts should have "context=trunkinbound" because trunkinbound is where the Vicidial script resides to capture and manage inbound calls. So: if your provider will be sending you calls from the same IP that you make your outbound calls ... all you need to do is add "context=trunkinbound" to the carrier settings and it will work IF the call can get to your server through the firewall.

Does your Vicidial server have a public IP address, a private IP address, or both?

[Cj4life24]

It has both a public and a private

[williamconley]

in which case accessing port 5060 is easy. do remember to close your iptables firewall to non-authorized sites/servers/users (ie: whitelist only access is safest).

[Cj4life24]

Ok,

so i have tried setting up the inbound but having a lot of trouble. I have followed the managers manual and added SALESLINE in group and created a did for the incoming numbers. I have made an inbound campaign and i think i have it all set up but when i call a number that is tied in with my sip trunk it is a long pause followed by a busy signal. What am i doing wrong the telco provider is sending me all 10 digits. Could you please help guide me i have a dead line of Monday Night to get this inbound working. Also sill haveing trouble accesing server from off site my firewall has exceptions for the proper ip addresses. I have also tried from inside the network to access the public address and i still cant get in. I have tried to putty or ssh i can connect to the 192 address but can not connest to the public address.

[Cj4life24]

Here is the Cli output for an incoming call it says pbx change but that is not the carrier it should be it should be telovations. I also get a message when call a number that it rings once and then the message says "the call could not be connected" or sometines after the first ring it goes busy.

[Nov 1 10:34:38] == Parsing '/etc/asterisk/manager.conf': [Nov 1 10:34:38] Found
[Nov 1 10:34:38] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 10:34:38] -- Executing [8600052@default:1] MeetMe("Local/8600052@default-8933,2", "8600052|F") in new stack
[Nov 1 10:34:38] > Channel Local/8600052@default-8933,1 was answered.
[Nov 1 10:34:38] -- Executing [914083709550@default:1] AGI("Local/8600052@default-8933,1", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 1 10:34:38] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 1 10:34:38] -- Executing [914083709550@default:2] Dial("Local/8600052@default-8933,1", "SIP/pbxchange/4083709550||tTor") in new stack
[Nov 1 10:34:38] -- Called pbxchange/4083709550
[Nov 1 10:34:40] -- SIP/pbxchange-0000886f is ringing
[Nov 1 10:34:40] -- SIP/pbxchange-0000886f is making progress passing it to Local/8600052@default-8933,1
[Nov 1 10:34:40] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 10:34:43] -- SIP/pbxchange-0000886f answered Local/8600052@default-8933,1
[Nov 1 10:34:49] == Parsing '/etc/asterisk/manager.conf': [Nov 1 10:34:49] Found
[Nov 1 10:34:49] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 10:34:49] -- Executing [58600052@default:1] MeetMe("Local/58600052@default-b147,2", "8600052|Fmq") in new stack
[Nov 1 10:34:49] > Channel Local/58600052@default-b147,1 was answered.
[Nov 1 10:34:49] -- Executing [8309@default:1] Answer("Local/58600052@defau lt-b147,1", "") in new stack
[Nov 1 10:34:49] -- Executing [8309@default:2] Monitor("Local/58600052@default-b147,1", "wav|20121101-103448_4083709550_305") in new stack
[Nov 1 10:34:49] -- Executing [8309@default:3] Wait("Local/58600052@default-b147,1", "3600") in new stack
[Nov 1 10:34:51] == Manager 'sendcron' logged off from 127.0.0.1

[GaD]

There's actually information missing. Fomr what you posted above there is only the outbound call, there is notheing refering to when the call comes in. Please try the test again (hopefully from a separate system) and post the output that you get.

[williamconley]

that is not the output from an incoming call, it is from an outgoing call.