similar to: [asterisk-user] gsmtolin_framein: Invalid GSM data

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

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

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

I park a call and instead of the parked extension being returned, I get silence and the log shows a bunch of the following messages WARNING[26220]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from (null) (320)? what does this mean? BTW these messages are intermittant. sometimes it works fine other times i get the above message Regards

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, 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",

Hi list! If I call a number from the phone of my wife, I get this warning: [Jun 15 20:50:18] WARNING[21921]: translate.c:206 framein: no samples for gsmtolin (more time per seconds). I didn't found any help in Google with this message... Someone wrote about "turning off silence suppression", that it's already turned off... I tried to change the settings for the users,

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

----- "Chandrakant Solanki" <solanki.chandrakant at gmail.com> escreveu: > Hello > > I have successfully compiled OSLEC for echo cancellation for DAHDI > channel. > > Is there any way to do echo cancellation for SIP Channel. > > Is any, please suggest me.?? > > Thanks in advance.. > > -- > Regards, > > Chandrakant Solanki Short

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

Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using "meetme list" command it shows Minus in activity column. Any Idea. >meetme list Conf Num Parties Marked Activity Creation Locked 54682 0002 N/A

Hi Can someone explain me what is the purpose for MeetMe Option 'o'.. If I defined 'o' with MeetMe option or If not defined with MeetMe option... What is the difference between these two if defined or not defined MeetMe 'o' option... -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts

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

Hello All, I am trying to install Asterisk 1.8.13.0 & dahdi-complete 2.5.1 & libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. I am using "SunOS solaris-server 5.11 11.1 i86pc i386 i86pc" on Virtual Box. -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was

Hi All. I have export some db parameter in /etc/bashrc as follows ... export DB_NAME=xyz export DB_IP=1x.1x.1x.1x export DB_PWD=dkjfaoi Now, I want use these all environment variable into /etc/asterisk/res_mysql.conf file. Is there any way to do this..?? -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file "conf-kicked" and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-00000000) 2] Normal User (e.g. SIP/8484-00000001) 3] Admin User (e.g.

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

I current have a working Asterisk 1.2.12 server with ODBC voicemail storage, realtime static maps for voicemail, sip and iax configuration files. Realtime extensions, etc. All works great. I have verified that this configuration works on my test server as well. Now I am trying to test the 1.4B3 version on the same test server, and all works well except for ODBC voicemail. I am using the same

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call