asterisk users - May 2009 - MeetMe not working with GSM codec?

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

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
>

can you look on this from your debug


   1. app_meetme.c:3030 find_conf: The requested confno is '12'?
   2.   == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23]
   DEBUG[6872]: config.c:1306 config_text_file_load: Parsing
   /etc/asterisk/meetme.conf
   3.   == Found
   4. [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
>
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