[asterisk-users] MeetMe not working with GSM codec?

Chris Maciejewski chris at wima.co.uk
Fri May 22 04:12:16 CDT 2009


Thanks Kinjal!

Missing sound files was the problem. There were no .gsm files in my
sounds directory. Despite console shows .slin, the actual files
required are .gsm.

Once I copied .gsm into /var/lib/asterisk/sounds everything works OK.

Regards,
Chris


2009/5/22 Kinjal Dixit <kinjal.dixit at gmail.com>:
> On an entirely unrelated note, do you have the gsm asterisk sounds
> installed?  Maybe that vm-*.slin files don’t exist.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris
> Maciejewski
> Sent: Friday, May 22, 2009 12:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] MeetMe not working with GSM codec?
>
> Hi Dhaval,
>
> The reason confno '12' is not found in meetme.conf is because I am
> using MySQL as realtime config backend.
> See few lines below there is:
>
> [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478
> mysql_reconnect: MySQL RealTime: Connection okay.
> [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:365 realtime_mysql:
> MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno =
> '12'
>
> My meetme.conf:
> [general]
> audiobuffers=32
> logmembercount=yes
> schedule=no
>
>
>
> 2009/5/22 DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>:
>> can you look on this from your debug
>>
>> app_meetme.c:3030 find_conf: The requested confno is '12'?
>>   == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]:
>> config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf
>>   == Found
>> [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a
> valid
>> conference
>>
>> its on line number 318
>>
>> it seems that you doesent specify valid conference number
>> can you post meetme.conf
>>
>> regards
>> Dhaval
>>
>>
>> On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski <chris at wima.co.uk>
> wrote:
>>>
>>> Hi,
>>>
>>> I am not sure if I am doing something wrong, but I can't get MeetMe to
>>> work with GSM codec (Asterisk 1.6.1 SVN r190371).
>>>
>>> My config files below:
>>>
>>> ---- sip.conf: ----
>>> [general]
>>> context=common
>>> canreinvite=no
>>> bindport=5060
>>> bindaddr=78.105.1.127
>>> disallow=all
>>> allow=alaw
>>> allow=gsm
>>> rtptimeout=600
>>> rtpholdtimeout=3600
>>> rtpkeepalive=30
>>> nat=no
>>> jbenable=yes
>>> tcpenable=no
>>> realm=dev-sip.wima.co.uk
>>>
>>> [10000]
>>> type=friend
>>> secret=test
>>> host=dynamic
>>> nat=yes
>>> --------------------------
>>>
>>> ----- extensions.conf: -----
>>> [common]
>>> exten => 501,1,MeetMe(12,MI)
>>> exten => 501,n,Hangup()
>>>
>>> exten => i,1,Hangup()
>>> exten => h,1,Hangup()
>>> exten => t,1,Hangup()
>>> ------------------------------------
>>>
>>> Everything works OK when ALAW is used - see
>>> http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just
>>> after starting MeetMe application - see http://pastebin.com/f78d04c95
>>> line 327.
>>>
>>> Is there a problem with MeetMe app or I need to adjust my configuration?
>>>
>>> Regards,
>>> Chris
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



More information about the asterisk-users mailing list