similar to: asterisk crashes!!!

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 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 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",

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

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

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

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

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

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

[I'm reposting this to the asterisk-users list, since it seems to be a bit more active.] Hello, I started messing with Asterisk few days ago, so my overall knoledge about it is still fairy superficial. I think I found an issue with MP3Player; it can be reproducted with this extension: exten => 6001,1,Answer exten => 6001,2,Background(blahblah) exten => 6001,3,Ringing exten =>

Hi, 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 the SIP

I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.11.0 [Jun 2 17:08:28] WARNING[21652]:

Hi, Anybody seen this before? (using a pre-compiled asterisk from the OBS on a sles11sp1) (I mean, i did the same with a 1.6 without any problem, but i need 1.8) after starting: kc3004:~ # /usr/sbin/safe_asterisk: line 145: 16133 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} Asterisk ended with

I've just made (on FreeBSD 4.10) cvs co asterisk gmake gmake install asterisk -vvvvvvc and got a new error message (was working ok before the cvs) [codec_gsm.so]Oct 6 17:23:16 WARNING[135192576]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/codec_gsm.so: Undefined symbol "Short_term_analysis_filteringx" Oct 6 17:23:16 WARNING[135192576]: loader.c:429 load_modules:

Hi, I am using asterisk 1.6.0.10 For debugging i set verbosity to 10 with asterisk -vvvvvvvvvvr.. now i am trying to set it lower but.. when i type asterisk -r it starts with Connected to Asterisk 1.6.0.10 currently running on asterisk1 (pid = 2408) Verbosity is at least 10 when i try set verobisty 1 or similar commands.. i think this command is obselete in 1.6 .. set verbose 1 No such command