[asterisk-users] g729 problem
Hi iam using asterisk 1.2 version I have purchased g729 license from Digium when iam making calls, iam getting this error ? Jun 25 14:41:45 NOTICE[4424]: frame.c:183 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end any help ram ___ --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
Re: [asterisk-users] g729 problem
Disable Voice Activity Detection ram wrote: Hi iam using asterisk 1.2 version I have purchased g729 license from Digium when iam making calls, iam getting this error ? Jun 25 14:41:45 NOTICE[4424]: frame.c:183 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end any help ram ___ --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
Re: [asterisk-users] g729 problem
On 6/25/07, Karl J. Vesterling [EMAIL PROTECTED] wrote: Disable Voice Activity Detection yes i have disabled at my eyebeam, still i see this error iam using 1.2.18 ram ___ --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
Re: [asterisk-users] g729 problem
Hi, I had exactly the same problem with trixbox and asterisk 1.2.11. That looks to be a bug in this release of asterisk. When using one channel with one license, I had the message Out of G.729 Encoder Licenses! (in /var/log/asterisk/full). And once the call finished, the command 'show g729' indicates the licence is still in use (but there is no active channel ...). I've upgraded to zaptel-1.9.1 and asterisk 1.2.12.1 and I have no more problem ! Benoit o o wrote: Hoping someone can point me in the right direction. I have the following setup: Trixbox latest (asterisk 1.2.11) DID thru IAX trunk (EXGN/Vitelity) Termination thru IAX trunk (EXGN/Vitelity) 3 GXP-2000 phones with firmware (1.1.0.16) 6 Digium g729 licenses I have no codecs specified per trunk or extension. In iax.conf and sip.conf, with disallows=all and allow=gsm, everything works fine. When I change it to allow=g729, I have problems with incoming calls, and calls from the extensions into the system (voicemail, etc). Outgoing calls work just fine (no transcoding). Incoming calls are automatically answered by the system attendant with a quick greeting, then dumped into a call queue. With g729 enabled, calls get answered, but no audio is heard. Same with phones trying to check voicemail. After hanging up, it doesn't release the codecs. 'show g729' gives: 6/0 encoders/decoders of 6 licensed channels are currently in use And it will stay like that until asterisk is restarted. I'm assuming its a transcoding issue with the system attendant/prompts, but I can't see how it would need more than one license per incoming call, much less six, and why the licenses aren't released after the call ends. And I haven't found a console command that actually lists the per license usage. Any help resolving this would be appreciated. I already purchased 3 more licenses than I theoretically need, just to throw $30 at the problem and save some hair, but that didn't work. Thanks in advance. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Benoît Mérouze _._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._. Groupe IPercom - The VoIP Enabling Company - http://www.ipercom.com Ingénieur RD - courriel : [EMAIL PROTECTED] Network Software Developer - mailto: [EMAIL PROTECTED] Tél. / Phone : +33 1 7269 9611 ._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._ Siège Social 43, rue Fessart 92100 Boulogne Billancourt RCS NANTERRE B 440 345 528 - Capital social: 100 000 € CE COURRIEL COMME LES DOCUMENTS EVENTUELLEMENT ASSOCIES SONT CONFIDENTIELS COUVERTS PAR LE SECRET PROFESSIONNEL THIS E MAIL AND ANY DOCUMENT POSSIBLY ATTACHED ARE CONFIDENTIAL AND COVERED BY THE PROFESSIONAL SECRECY ._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._ Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Benjamin Franklin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 problem
On Thu, 2006-09-14 at 11:05 +0200, Benoît Mérouze wrote: Hi, I had exactly the same problem with trixbox and asterisk 1.2.11. That looks to be a bug in this release of asterisk. When using one channel with one license, I had the message Out of G.729 Encoder Licenses! (in /var/log/asterisk/full). And once the call finished, the command 'show g729' indicates the licence is still in use (but there is no active channel ...). I've upgraded to zaptel-1.9.1 and asterisk 1.2.12.1 and I have no more problem ! The issue of one way audio with g.729 on Trixbox with the (then) latest Asterisk packages that the original author reported is not caused by a bug in Asterisk itself. Afaik it's caused by a shortcoming in the Digium g.729 codec. Apparently the g.729 codec does not have it's own PLC which causes one way audio if they are used with an Asterisk release that uses the SIP JitterBuffer from http://www.asterisk-backports.org. Trixbox Asterisk packages use this SIP JitterBuffer so have the issue when using g.729. The latest packages as seen on http://www.laimbock.com/asterisk/ have a solution for this problem by offering two different Asterisk rpms where one of them does not have the SIP JitterBuffer patch applied. So if you want the latest and use g.729, get the packages with NoJB in the name and you should be ok. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 problem
o o wrote: Thomas, Thanks for your help so far. I finally figured out where 'debug level 10' dumps to. In reading the logs there, it's telling me I'm out of licenses. I'm not a math wizard by any means, but I would assume with g729 on the GXP-2000 and on the IAX trunk, I would only need 1 license to transcode my IVR prompts to the incoming caller. Well, first-off it would be a good idea to re-encode them to g.729. If you are using trunk then there is a convert tool for this, failing that you can use the tool on. http://www.asteriskguru.com/tools/audio_conversion.php The ast-linux site can provide you with all the default prompts re-encoded (well, re-recorded then encoded). However, it seems to use all 6 available, and never releases them, even after hanging up the call. I haven't found a way to see what process(s) are using each license instance. If you are using monitor/mixmonitor or a meetme room you will run out of licenses very quickly as both applications will need to transcode the stream to slin and bakc again. I downloaded the re-recorded set you referenced, and I can hear the default system prompts, but all my previously recorded prompts are null because of the 'out of license' issue. For the recording, even with console verbosity set to 16, the out of license messsage was never logged to the console, Strange, maybe this behaviour changed in later versions (It's been a while shice it happened to me). only to the debug text. I'm off to find a way to transcode my custom prompts into g729 (or get the freepbx recording interface to do so for new prompts) See above. but if someone can help me determine why * thinks I need more than 6 licenses for a single incoming call I would appreciate it. TIA Which applications are running alongside the call? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g729 problem
Hoping someone can point me in the right direction. I have the following setup: Trixbox latest (asterisk 1.2.11) DID thru IAX trunk (EXGN/Vitelity) Termination thru IAX trunk (EXGN/Vitelity) 3 GXP-2000 phones with firmware (1.1.0.16) 6 Digium g729 licenses I have no codecs specified per trunk or extension. In iax.conf and sip.conf, with disallows=all and allow=gsm, everything works fine. When I change it to allow=g729, I have problems with incoming calls, and calls from the extensions into the system (voicemail, etc). Outgoing calls work just fine (no transcoding). Incoming calls are automatically answered by the system attendant with a quick greeting, then dumped into a call queue. With g729 enabled, calls get answered, but no audio is heard. Same with phones trying to check voicemail. After hanging up, it doesn't release the codecs. 'show g729' gives: 6/0 encoders/decoders of 6 licensed channels are currently in use And it will stay like that until asterisk is restarted. I'm assuming its a transcoding issue with the system attendant/prompts, but I can't see how it would need more than one license per incoming call, much less six, and why the licenses aren't released after the call ends. And I haven't found a console command that actually lists the per license usage. Any help resolving this would be appreciated. I already purchased 3 more licenses than I theoretically need, just to throw $30 at the problem and save some hair, but that didn't work. Thanks in advance. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 problem
o o wrote: Hoping someone can point me in the right direction. I have the following setup: It is worth noting, that if you have a console open and you run out of licenses, (I don't know at which verbose level this is) You are made very aware of it. IIRC roughly 10 messages a second warning you t hat you have run out of licenses. Also, it is a very good idea to re-encode any voice prompts etc. into g729 (if you are looking for transcoded defaults, a rerecorded set can be downloaded from astlinux.org in the AstLinux, Downloads, Extras, Sounds section). As for the original problem, have you looked at the console output with the debug level set high? Have you looked at any debug/syslog output from the Handsets? Have you captured any debug output from sip debug . etc.? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 problem
--- Thomas Kenyon [EMAIL PROTECTED] wrote: o o wrote: Hoping someone can point me in the right direction. I have the following setup: It is worth noting, that if you have a console open and you run out of licenses, (I don't know at which verbose level this is) You are made very aware of it. IIRC roughly 10 messages a second warning you t hat you have run out of licenses. Also, it is a very good idea to re-encode any voice prompts etc. into g729 (if you are looking for transcoded defaults, a rerecorded set can be downloaded from astlinux.org in the AstLinux, Downloads, Extras, Sounds section). As for the original problem, have you looked at the console output with the debug level set high? Have you looked at any debug/syslog output from the Handsets? Have you captured any debug output from sip debug . etc.? Thomas, Thanks for your help so far. I finally figured out where 'debug level 10' dumps to. In reading the logs there, it's telling me I'm out of licenses. I'm not a math wizard by any means, but I would assume with g729 on the GXP-2000 and on the IAX trunk, I would only need 1 license to transcode my IVR prompts to the incoming caller. However, it seems to use all 6 available, and never releases them, even after hanging up the call. I haven't found a way to see what process(s) are using each license instance. I downloaded the re-recorded set you referenced, and I can hear the default system prompts, but all my previously recorded prompts are null because of the 'out of license' issue. For the recording, even with console verbosity set to 16, the out of license messsage was never logged to the console, only to the debug text. I'm off to find a way to transcode my custom prompts into g729 (or get the freepbx recording interface to do so for new prompts) but if someone can help me determine why * thinks I need more than 6 licenses for a single incoming call I would appreciate it. TIA __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 problem HELP!
Dear i have buy two license of G729 codec and i have install/registered as documented but after i start Asterisk -vvvcng i notice this warning and if i made call the CLI say No compatible codec! How can i solve this problem? Thanks in advance Dimitri -- [app_datetime.so] = (Date and Time) == Registered application 'DateTime' [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 2 licensed G.729 transcoders Apr 21 20:52:15 WARNING[16384]: translate.c:213 calc_cost: Translator 'g729tolinb' does n t produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 43 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI Apr 21 20:52:17 NOTICE[131081]: chan_sip.c:5880 sip_poke_noanswer: Peer 'santoext' s now UNREACHABLE! Apr 21 20:52:36 WARNING[131081]: chan_sip.c:2113 process_sdp: No compatible codecs! Apr 21 20:52:38 NOTICE[131081]: chan_sip.c:5337 handle_request: Failed to authenticate us r sip:[EMAIL PROTECTED]:5060;tag=c398050c4b92f090 *CLI -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
