search for: gsmtolin

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, allowing just ulaw and alaw, but it's the same... Can someone...

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 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 samp...

...\371\222\327&X\244\357\221$J\332^A\233\372\331\343$^L+L\362\333\204D\234\243Tt\325\227F\375\221\223\243G^_\211-\213\335\243I=,\354\\225d\263'\212I"\341#\207oj\272\352\325^T\206\275D\343$\214\363\262KQ\233^D^K#u\334\225\352\242\355\\227\346[Feb 24 16:07:20] WARNING[777]: codec_gsm.c:140 gsmtolin_framein: Invalid GSM data (1) [Feb 24 16:07:20] WARNING[777]: translate.c:197 framein: gsmtolin did not update samples 0 [Feb 24 16:07:20] WARNING[777]: codec_gsm.c:140 gsmtolin_framein: Invalid GSM data (1) [Feb 24 16:07:20] WARNING[777]: translate.c:197 framein: gsmtolin did not update samples 0...

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 [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 samp...

Our office does not have a PA system, and in our current phone system we have a certain extension that we dial that pages over the speaker of all the phones in the office. Does Asterisk support this feature? If so, could someone tell me the best way to set this up in AsteriskNOW? Thanks, Nick -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ox 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 translator is loaded... [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 4 [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 1b:a1:18:82:47:6f:a8:f7:33:de:7d:77:e8:1d:60:15:53:ec:49:aa == Found license &...

...=> (Comedian Mail (Voicemail System) with ODBC Storage) == Registered channel type 'Local' (Local Proxy Channel Driver) chan_local.so => (Local Proxy Channel) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_gsm: using generic PLC == Registered translator 'gsmtolin' from format gsm to slin, cost 5 asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: odbc_request_obj I get no other information in the debug or message files. An attempt to backtrace, does not yield a crash dump regardless of the compile options. Does...

...s I forced it to G.729. For some reason incoming and outgoing calls will ALWAYS use G.711a. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The translator is loaded... [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 4 [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 1b:a1:18:82:47:6f:a8:f7:33:de:7d:77:e8:1d:60:15:53:ec:49:aa == Found license &...

...CM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 16 == Registered translator 'lintoilbc' from format slin to ilbc, cost 90 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format gsm to slin, cost 5 == Registered translator 'lintogsm' from format slin to gsm, cost 13 [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 13 == Registered translator '...

Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060512/98a6f962/attachment.htm

Hello! Asterisk "CVS-HEAD-08/06/04-14:55:13" won't start on two of three different Gentoo machines. This is the output of gdb: ultra asterisk # gdb /usr/sbin/asterisk GNU gdb 6.0 Copyright 2003 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions.

...> (ITU G.726-32kbps G726 Transcoder) == Registered translator 'g726tolin' from format G726 to SLINR, cost 99 == Registered translator 'lintog726' from format SLINR to G726, cost 77 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 86 == Registered translator 'lintogsm' from format SLINR to GSM, cost 85 [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Ouch ... error while writing audio data: : Broken pipe [root@localhost alix]# Thanks in advance for your hel...

...zapbarge.so] => (Barge in on Zap channel application) == Registered application 'ZapBarge' [app_sendtext.so] => (Send Text Applications) == Registered application 'SendText' [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format gsm to slin, cost 1 == Registered translator 'lintogsm' from format slin to gsm, cost 1 [app_image.so] => (Image Transmission Application) == Registered application 'SendImage' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Lo...

...m format ilbc to slin, cost 28 == Registered translator 'lintoilbc' from format slin to ilbc, cost 146 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Parsing '/etc/asterisk/codecs.conf': Not found (No such file or directory) == Registered translator 'gsmtolin' from format gsm to slin, cost 5 == Registered translator 'lintogsm' from format slin to gsm, cost 19 [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Parsing '/etc/asterisk/codecs.conf': Not found (No such file or directory) == Registered translator...

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in