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

[Chris Maciejewski]

Hi Martin,

Yes, I do have GSM compiled for sure.

$asterisk -r -x "core show codecs audio"

Disclaimer: this command is for informational purposes only.
	It does not indicate anything about your configuration.
        INT    BINARY        HEX   TYPE       NAME   DESC
--------------------------------------------------------------------------------
          1 (1 <<  0)      (0x1)  audio       g723   (G.723.1)
          2 (1 <<  1)      (0x2)  audio        gsm   (GSM)
          4 (1 <<  2)      (0x4)  audio       ulaw   (G.711 u-law)
          8 (1 <<  3)      (0x8)  audio       alaw   (G.711 A-law)
         16 (1 <<  4)     (0x10)  audio   g726aal2   (G.726 AAL2)
         32 (1 <<  5)     (0x20)  audio      adpcm   (ADPCM)
         64 (1 <<  6)     (0x40)  audio       slin   (16 bit Signed Linear PCM)
        128 (1 <<  7)     (0x80)  audio      lpc10   (LPC10)
        256 (1 <<  8)    (0x100)  audio       g729   (G.729A)
        512 (1 <<  9)    (0x200)  audio      speex   (SpeeX)
       1024 (1 << 10)    (0x400)  audio       ilbc   (iLBC)
       2048 (1 << 11)    (0x800)  audio       g726   (G.726 RFC3551)
       4096 (1 << 12)   (0x1000)  audio       g722   (G722)


I will open a bug report.

Regards,
Chris

2009/5/22 Martin <asterisklist at callthem.info>:
> it should work just fine; do you have the GSM codec compiled/loaded ????
>
> core show modules like codec_gsm ... ?
>
> OR that particular version has a BUG...
>
> Martin
>
> On Thu, May 21, 2009 at 3:56 AM, 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
>