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@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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Posted: Fri May 22, 2009 6:29 am Post subject: [asterisk-users] MeetMe not working with GSM codec?
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@callthem.info>:
Quote:
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@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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Posted: Fri May 22, 2009 6:52 am Post subject: [asterisk-users] MeetMe not working with GSM codec?
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@lists.digium.com
[mailto:asterisk-users-bounces@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
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
Quote:
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@wima.co.uk>
wrote:
Quote:
>
> 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 --
Posted: Fri May 22, 2009 7:21 am Post subject: [asterisk-users] MeetMe not working with GSM codec?
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@wima.co.uk> wrote:
Quote:
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@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@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 --
Posted: Fri May 22, 2009 8:18 am Post subject: [asterisk-users] MeetMe not working with GSM codec?
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@gmail.com>:
Quote:
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@lists.digium.com
[mailto:asterisk-users-bounces@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@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@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
>
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