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

Martin asterisklist at callthem.info
Fri May 22 03:17:00 CDT 2009


this command doesn't show the codecs present in the system .... do you
have g723 compiled too ?
try core show translations or something like that

Martin

On Fri, May 22, 2009 at 2:25 AM, Chris Maciejewski <chris at wima.co.uk> wrote:
> 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
>>>
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>>
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>
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