similar to: what the heck? codec_gsm.c:135 gsmtolin_framein: Huh?

Hi guys, Anyone seen something like below(see below the line)? Machine P2 w/512MB RAM Debian (testing) ; kernel 2.6.12-1-386 asterisk 1.2.1-n-all incl. astcc For many months now I went through * 1.07, 1.09 and never saw something like that. Even with 1.2.0, a month now, at the beginning everything was fine, and suddenly "codec_gsm.c:194 gsmtolin_framein: Invalid GSM data" thing

Hi I have created meetme with 3 user. When i going to mute user it gives following error.. *Asterisk Version : 1.6.2.6* -- <SIP/52987-00000040> Playing 'conf-muted.gsm' (language 'en') [Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid GSM data (1) [Jul 2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not update samples 0 [Jul

Hi, I got ast. 1.6.0.10 working for a few weeks without a problem. A few mins ago..I got the following msgs on ast-cli and asterisk service crashed. I coudlnt find anything that might cause this problem. Any ideas?? [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0

Apparently asterisk cannot reboot gracefully (unattended) when using realtime.... MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. WARNING[3763]: MySQL RealTime: Couldn't establish connection. Check debug. Since asterisk starts before mysql it will never start any workaround for this? g Regards Greg Cirino

I have installed G729 but I cannot make a outgoing call with it. SIP/dennis-2c23 is making progress passing it to SIP/1010-8b60 NOTICE[311316]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from G729A to ALAW NOTICE[311316]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from GSM to G729A WARNING[311316]: File codec_gsm.c, Line 136

Hi, I just installed asterisk 1.4.0 on my mac. I compiled from source with no issues. I installed the sample config files, and basically just added a register line to sip.conf (to register with a Free World Dialup account). Then I called my asterisk system from a different computer (using x-lite softphone on windows xp, registered to an ekiga.net account). Asterisk answers, and I can hear the

I am trying to resolve a problem where grandstream phones (only) fail to register after a period of time. I have a mysql realtime setup that appears to work, but fails for no reason. before i classify the realtime system as unusable, I am trying to isolate the problem. One thing I have noticed is the odbc driver continues to load even though I have removed (from what I can tell) references to

It's been hours since I've seen a post from this list Must be broken again. Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You

Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial("Zap/2-1",

Has anybody been able to get in touch with anybody at digium today? Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following

does anybody have an idea what the difference and significance of sip seeding and registration is. g Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX

I am getting thousand of these messages in asterisk console Jun 27 12:35:55 WARNING[16496]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data And after some time the system crashes. Does anyone know why? I running Asterisk SVN-trunk-r7522 built Does it help to upgrade the system? Regards, Fredrik Jensen

Is there some reason the sipbuddies table structure was designed with sip config values as column names? Doesn't look very flexible It really should take the form of ast_config so when a new sip feature is implemented, you don't have to re-write the entire data structure too. Regards Greg Cirino

it appears the cvs for zaptel as of 12/13/04 am is missing at least 1 file -- wcfxs.c greg Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When

Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to

Hi, Has anybody had any problems getting digium hardware lately? Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

Hi everyone, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange. Anyhow I was getting an error: Process_sdp: No compatible codecs! And from