vicidial.org

Posted: **Sat Sep 22, 2007 9:41 am**

I have installed asterisk 1.2.17 with the help of Scratch Install on a AMD Athlon 3000+ 1.8Mhz Machine. I have got registered with the Asterisk Box and I could call inhouse(loacally)as well as outbound Distant calls to UK when i tried direct Call from Eye Beam(eyeBeam 1.1 3010n stamp 19039). But When i logged into the vicidial, there was no incoming call from asterisk.
I have noticed that there is no perl script running for AST_manager_send.pl and AST_manager_listen.pl,

when i tried ps-e | grep AST

So i tried to start the sscript, pearl AST_manager_send.pl in "/usr/src/astguiclient/bin" directory and i got this output

[root@localhost bin]# perl AST_manager_send.pl
checking to see if listener is dead ||0|
LISTENER DEAD STOPPING PROGRAM... ATTEMPTING TO START keepalive SCRIPT
loop counter: |863989|
PROCESS KILLED MANUALLY... EXITING

checking to see if listener is dead |1|0|
LISTENER DEAD STOPPING PROGRAM... ATTEMPTING TO START keepalive SCRIPT
DONE... Exiting... Goodbye... See you later... Not really, initiating next loop...0 left
DONE... Exiting... Goodbye... See you later... Really I mean it this time

and when i run the pearl AST_manager_send.pl in "/usr/src/astguiclient/bin" directory and i got this output

[root@localhost bin]# unknown remote host: at AST_manager_listen.pl line 199
-bash: unknown: command not found

I have tried to call "8600001" from eyebeam, the call got connected and there was pure silence and i have noticed that in the there was an asterisk log saying

[root@localhost ~]# asterisk -r

Connected to Asterisk 1.2.17 currently running on localhost (pid = 8782)
Verbosity is at least 20
-- Registered SIP '3001' at 192.168.2.254 port 5060 expires 3600
-- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 3001
-- Executing MeetMe("SIP/3001-09cfccb8", "8600001|q") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600001'
Sep 22 19:56:45 WARNING[8897]: channel.c:2385 set_format: Unable to find a codec translation path from g729 to slin
Sep 22 19:56:45 WARNING[8897]: app_meetme.c:989 conf_run: Unable to set 'SIP/3001-09cfccb8' to write linear mode
-- Hungup 'Zap/pseudo-795110115'
== Spawn extension (default, 8600001, 1) exited non-zero on 'SIP/3001-09cfccb8'
-- Executing DeadAGI("SIP/3001-09cfccb8", "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 DeadAGI("SIP/3001-09cfccb8", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0---------------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0
-- Incoming call: Got SIP response 500 "Server Internal Error" back from 192.168.2.254
localhost*CL

Somebody please help me in this as i have reinstalled it more than 5 times, and that leaded me to sleepless night.

:cry:

Looking Forward

Benoy

Posted: **Sat Sep 22, 2007 2:51 pm**

ep 22 19:56:45 WARNING[8897]: channel.c:2385 set_format: Unable to find a codec translation path from g729 to slin
Sep 22 19:56:45 WARNING[8897]: app_meetme.c:989 conf_run: Unable to set 'SIP/3001-09cfccb8' to write linear mode
-- Hungup 'Zap/pseudo-795110115'

you dont have g729

purchase one from digium.

thanks.

or use alaw and ulaw.

Posted: **Sun Sep 23, 2007 1:12 am**

Thanks for your reply and co-operation,

okie i will do that.

But what would be the problem of no incoming call when i login to vicidial.. is that becuse of the codec issue...?

Posted: **Sun Sep 23, 2007 1:43 am**

Hi

thats correct due to codec issue you wont be able to establish the call

so try to fix in your sip config use ulaw and try.

ram

Posted: **Mon Sep 24, 2007 5:44 am**

I made that to ulaw... No errors/warnings when i call from Direct Eyebeam.

*CLI> -- Executing AGI("SIP/3003-b7a06188", "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 Monitor("SIP/3003-b7a06188", "wav|20070924-155512_3003_44774944 8585") in new stack
-- Executing Dial("SIP/3003-b7a06188", "SIP/00447749448585@100213|100|tTo") in new stack
-- Called 00447749448585@100213
-- SIP/100213-08dd4ed8 is making progress passing it to SIP/3003-b7a06188
-- SIP/100213-08dd4ed8 answered SIP/3003-b7a06188
== Refreshing DNS lookups.
== Refreshing DNS lookups.

But even now I am not getting the incoming call saying, "you are the only person in this conference" when i log into vicidial. I think that cluld be because of perl scripts.

:cry:

Then I tried to call 8600001.. which got connected but gave me pure silence. And there was no errors in th log as well.

i have noticed that No AST_manager_listen.pl, AST_manager_send.pl running when tried

ps -e | grep AST

And I have one more doubt, what is screen -r...
i have tried screen -r

[root@localhost ~]# screen -r
There are several suitable screens on:
5075.ASTVDauto (Dead ???)
4180.ASTVDauto (Detached)
3331..localhost (Dead ???)
3215.ASTVDadapt (Dead ???)
3218.ASTfastlog (Dead ???)
8644..localhost (Dead ???)
5131..localhost (Dead ???)
2899.ASTfastlog (Detached)
8941..localhost (Dead ???)
2896.ASTVDadapt (Detached)
14903..localhost (Dead ???)
7850..localhost (Dead ???)
Remove dead screens with 'screen -wipe'.
Type "screen [-d] -r [pid.]tty.host" to resume one of them

Actually i dont klnow what does it mean..

My core problem is no incoming call from asterisk box when we login to vicidial which says

"you are the only person in this conference"

Looking Forward,

Thanks a lot for ur helps and outlooks

Posted: **Mon Sep 24, 2007 6:16 am**