Problem With Login in ViciDial - No incoming Call to EyeBeam

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Problem With Login in ViciDial - No incoming Call to EyeBeam

Postby benoyantony » Sat Sep 22, 2007 9:42 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

Quote:

[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


Quote:

[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

Quote:


[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.



Looking Forward

Benoy
benoyantony
 
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Postby ramindia » Sat Sep 22, 2007 2:37 pm

Hi

did you followed Scartch_install CLOSER Section

next thins i see some error codec translation

Sep 22 19:56:45 WARNING[8897]: channel.c:2385 set_format: Unable to find a codec translation path from g729 to slin


what codec at server side and client side you are using

ram
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Postby benoyantony » Sun Sep 23, 2007 1:20 am

I am using G729 codec in both client and server side
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Postby ramindia » Sun Sep 23, 2007 1:44 am

Hi

Do not post same problem twise.

check other post some suggestions

ram
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Postby benoyantony » Sun Sep 23, 2007 9:53 am

It was by mistake.. was trying to put with an email id and it was not supporting as I am new to the Forum. Am so sorry to bother you..
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Postby gardo » Sun Sep 23, 2007 11:55 am

make sure you have the g729 codec installed in your asterisk. then check your crontab entries and compare it w/ the vicidial scratch install. running "show translation" w/in the asterisk cli will show you the installed codecs.
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Postby Op3r » Sun Sep 23, 2007 5:56 pm

the proper command to check your g729 license in asterisk is show g729

remember using g729 will take a toll on your server. Please plan accordingly.
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Postby benoyantony » Mon Sep 24, 2007 6:18 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
benoyantony
 
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Joined: Sat Sep 22, 2007 9:03 am

Postby benoyantony » Mon Sep 24, 2007 6:21 am

this is the output i got
ps -e | grep AST

[root@localhost ~]# ps -e | grep AST
2897 pts/5 00:00:08 AST_VDadapt.pl
8013 ? 00:00:00 AST_manager_kil
[root@localhost ~]#



Sometimes its showing AST_AutoDial... also
benoyantony
 
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Postby ramindia » Mon Sep 24, 2007 6:25 am

Hi

screen -wipe

then paste screen -r

check the crontab and run them manually and see what are the errors you will see.

ram
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so its very usefull for others who join later as a NEWBIE.
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Postby benoyantony » Mon Sep 24, 2007 10:05 am

[root@localhost ~]# screen -wipe
There are screens on:
5887.ASTVDauto (Detached)
3660.pts-1.localhost (Attached)
2338.ASTVDadapt (Detached)
2340.ASTfastlog (Detached)
4 Sockets in /tmp/screens/S-root.

[root@localhost ~]# screen -r
There are several suitable screens on:
5887.ASTVDauto (Detached)
3660.pts-1.localhost (Attached)
2338.ASTVDadapt (Detached)
2340.ASTfastlog (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.


This is the results when i got 'screen -r' after 'screen -w'

I think some perl scripts are not working...

I dont get the meaning of the following lines in scract install.. and i never run any command related to screen other than "screen -r","screen -d" and "screen -w" with out any other option or parameter

- once your system starts up you can attach to the screen running asterisk by
typing "screen -r " find which screen by typing "screen -r" and
looking for the lowest screen number. Then to detach again from the screen
while you are in it type 'Ctrl+a' then 'd'
- you are done
benoyantony
 
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Postby mflorell » Thu Sep 27, 2007 8:35 am

make sure your manager.conf is set up properly and that Asterisk is running. Your manager-connected scripts are not running.
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