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 Post subject: cid-callback
PostPosted: Mon Jun 07, 2010 4:03 pm 
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Joined: Sun Apr 11, 2010 11:20 am
Posts: 13
Hello,
i have installed the callback module and it's working fine if i use the customer web interface. but i want to use the cid-callback context so the user just makes a call to the DID and the system calls him again to enter the destination he wants to call. i got a DID and the calls now are forwarded to the system.
am putting this context:
[a2billing-cid-callback]
exten => _X.,1,AGI(a2billing.php,1,cid-callback|) ;last #parameter is the callback area code
exten => _X.,n,Hangup
when i try to call the DID it's asking for the PIN while am adding my caller ID to an account in the database and am expecting it to call me after i hangs up. please can u tell me what's missing in my setup.
thnx


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 Post subject: Re: cid-callback
PostPosted: Tue Jun 08, 2010 7:07 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4064
Hi

You are using AGI, you should be using deadagi.

In respect of your dial plan, you are using pipes and commas.

Code:
exten => _X.,1,AGI(a2billing.php,1,cid-callback|) ;last #parameter is the callback area code


If you are using Asterisk 1.6, use commas, if using 1.4, use pipes.

Joe


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 Post subject: Re: cid-callback
PostPosted: Tue Jun 08, 2010 8:15 am 
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Joined: Sun Apr 11, 2010 11:20 am
Posts: 13
hi Joe,
i put the dialplan as u said and now it's taking the correct caller ID and authenticating it fine and also the callback is triggered to initiate the callback. but after that am getting a strange error on the asterisk. below is the debug. am using asterisk 1.6.
am not sure what's making the asterisk crashing although i have recompiled it and still the same error.

-- Executing [[email protected]:1] AGI("SIP/vitel1-00000004", "a2billing.php,1,cid-callback,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- <SIP/vitel1-00000004>AGI Script a2billing.php completed, returning -1
VLYNX-RETAIL*CLI> *** glibc detected *** /usr/sbin/asterisk: free(): corrupted unsorted chunks: 0x00002aaac8006a20 ***
======= Backtrace: =========
/lib64/libc.so.6[0x32bbc7230f]
/lib64/libc.so.6(cfree+0x4b)[0x32bbc7276b]
/usr/sbin/asterisk(ast_cdr_detach+0x20d)[0x43f68d]
/usr/sbin/asterisk(ast_hangup+0x402)[0x448242]
/usr/sbin/asterisk[0x4ad065]
/usr/sbin/asterisk[0x4adf3b]
/usr/sbin/asterisk[0x4e383c]
/lib64/libpthread.so.0[0x32bc40673d]
/lib64/libc.so.6(clone+0x6d)[0x32bbcd3d1d]
======= Memory map: ========


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 Post subject: Re: cid-callback
PostPosted: Tue Jun 08, 2010 8:19 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4064
Hi

AGI("SIP/vitel1-00000004", "a2billing.php,1,cid-callback,")

use deadAGI, although this will probably not solve your crashing issues.

When recompiling, try with make distclean, then make clean to ensure you start with a clean installation.

Joe


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 Post subject: Re: cid-callback
PostPosted: Tue Jun 08, 2010 8:34 am 
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Joined: Sun Apr 11, 2010 11:20 am
Posts: 13
Hi Joe,
i have used the deadAGI and recompiled using the cleandist and now i got a different error as u see below:
Executing [[email protected]:1] DeadAGI("SIP/vitel1-00000000", "a2billing.php,1,cid-callback,") in new stack
[Jun 8 01:21:15] WARNING[17666]: res_agi.c:3081 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases!
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- <SIP/vitel1-00000000>AGI Script a2billing.php completed, returning -1
ServerA*CLI>
Disconnected from Asterisk server
/usr/sbin/safe_asterisk: line 139: 17618 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
Asterisk ended with exit status 139
[[email protected] asterisk]# Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed


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 Post subject: Re: cid-callback
PostPosted: Tue Jun 08, 2010 8:38 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4064
Hi

Ignore the deadagi warning.

in respect of the Asterisk problems, you may get better advice from the asterisk forums

Joe


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 Post subject: Re: cid-callback
PostPosted: Tue Jun 15, 2010 11:11 am 
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Joined: Sun Apr 11, 2010 11:20 am
Posts: 13
Hello,
i have installed the asterisk 1.4 and now iam no longer facing this problem. the system is calling me back and asking me to enter the destination number i wish to call. but when i start typing the digits it's giving busy tone right after like 5 seconds. regarding the route am using it supports DTMF 100%. what other things i have to check? any idea?


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