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Could Not Set Write Format To Slinear


Cancelling call to 1818335xxxx[Dec 12 07:53:20] VERBOSE[25372] logger.c: -- Couldn't call aretta/1818335xxxx[Dec 12 07:53:20] VERBOSE[25372] logger.c: == Everyone is busy/congested at this time (0:0/0/0)[Dec 12 07:53:20] DEBUG[25372] app_macro.c: Executed application: Dial[Dec Do a core show translation and make sure that you have a g.729 to slin path. admin2all 2013-05-30 16:33:36 UTC #18 I did the download and untarred the codec to modukles dir. But I don't know for sure.

The g.729 is a Digium product, they support it. Interesting fact, is that if I change the Inbound Destination from Queue to any extension it rings fine. I think it is a format problem because I get this: [Nov 11 11:57:10] VERBOSE[6250] file.c: -- Playing 'beep.g729' (language 'es')[Nov 11 11:57:11] WARNING[6250] channel.c: Unable to find a codec translation You indeed have codec errors in that log and if you absolutely must use G.729 you must buy licenses from Digium as there are still active patents on that codec...

Unable To Find A Codec Translation Path From (ulaw) To (g729)

I get similar output in my CLI - talking about codec translations etc. Should it matter if your Settings/Asterisk Sip Settings and extension config make the same disallow and allow settings? [2014-09-30 10:11:31] VERBOSE[9976][C-0000000a] pbx.c: -- Executing [[email protected]:1] Answer("PJSIP/780-00000006", "") in new stack[2014-09-30 10:11:31] My specific problem is when I make an outbound call to a 3rd party or an extension call to my other office ( they have their own setup like this one We could make and get phone calls and see the codec being used.

matsj 2012-02-07 13:08:00 UTC #3 it says speex followed by dashes -no translation/transcoding will be done? I have a Trunk conncetion (using g729) which I filled down as destination a Queue. No calls are possible with this config. Unable To Find A Codec Translation Path From 0x100 (g729) To 0x40 (slin) This is what I get...0/0 encoders/decoders of 20 licensed channels are currently in use Licenses Found:File: G729-XXXXXXXXX.lic -- Key: G729-XXXXXXXXXX -- Host-ID: d3:25:16:e0:d3:c3:cc:84:25:c0:9d:d6:9f:c2:67:18:e7:54:38:0a -- Channels: 20 (Expires: 2032-02-19) (OK) freepbx-voip*CLI> sip

Normally I forward a range of ports in my router but, whilst I was doing a set up on the router, I inadvertently deselected the forwarding. Asterisk Install G729 Our present in-use system has the same number of licenses 20. admin2all 2013-05-29 15:02:53 UTC #10 Hello Skyking - I downloaded the g729 from the Admin Panel drop down - under Digium Addons. Unable to find a codec translation path from (g729) to (slin) and think that it's problem the system is having using codecs.

Have a nice day! Asterisk Hosting Lv I have set Disallow to ALL and Allow to g729&ulaw&alaw&gsm. Skip to content Wiki Blog Forums Mailing Lists Contact Us Advanced search Forums have moved to https://community.asterisk.org Board index RSS RSS Change font size FAQ Information The requested topic does not Also check that permissions are 755 put it in/usr/lib/asterisk/modulesfor 64bit and if all the other asterisk modules are there it goes in /usr/lib64/asterisk/modules mustardman 2013-05-29 22:46:56 UTC #15 You download it

Asterisk Install G729

I am guessing it's a config issue with the g729. Among the Addons there was an option for the g729 codec which seemed to install fine. Unable To Find A Codec Translation Path From (ulaw) To (g729) Referenced by __ast_play_and_record(), and ast_dtmf_stream(). { struct ast_silence_generator *state; if (!(state = ast_calloc(1, sizeof(*state)))) { return NULL; } state->old_write_format = chan->writeformat; if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) { ast_log(LOG_ERROR, "Could not set Freepbx G729 Has someone any ideas?

Home Categories FAQ/Guidelines Terms of Service Privacy Policy Powered by Discourse, best viewed with JavaScript enabled Log In Problem with Queue called direct from Inbound General Help rafael_sfc 2014-02-19 14:28:04 UTC Could you provide a link? dicko 2013-05-15 19:43:53 UTC #3 . . . I only have 3 test phones hooked up to my system. Asterisk G729 Codec

Under Settings/Asterisk Sip Settings I have ulaw and gsm as the only allowed codecs. Somehow Freepbx is causing the conflict to occur. If this is not a Codec issue as its suggested here, can someone provide options or give me some set up help? g729 does not even show up on a core show translation..

I even entered that key in the GUI on the server !! Asterisk Show Codecs In Use gene778 2014-10-01 14:28:41 UTC #3 Thanks for the reply. I don't have access to my router right now but I think the range is normally 8000 to 26000.

Thank you a lot!

For outbound it's a fast busy. admin2all 2013-05-16 18:03:02 UTC #6 I don't fully understand what is supposed to happen, but when I move g729 codec down one space in the Asterisk SIP Settings I CAN make I smell something context related, mainly Freepbx queue context with from-trunk. Freepbx Codecs Is there a way I can force use g729 not matter what kind of call is made - IAX, Internal, Outgoing, etc,etc.??

I am running across an issue that I am at an impasse on and hope you can help/guide me.. I also see files like format_ulaw.so etc - and there is a file called format_g729.so !! Help !! The problem is about the codec?

that Asterisk will concurrently handle the audio on. Thanks...Andrew Log... I will contact them. IraHolden 2011-11-11 14:21:25 UTC #2 In General Settings, see if the Call Recording Format is set to g729.If so, try WAV.

Parameters: chanThe channel to generate silence on Returns:An ast_silence_generator pointer, or NULL if an error occurs This function will cause SLINEAR silence to be generated on the supplied channel until it admin2all 2013-05-16 13:05:49 UTC #4 I'm confused. surfparadise 2016-01-16 15:13:13 UTC #3 I didn't buy any codec... I applied our license and tested the system.

It should show up, even if not installed. You say that g729 commands show up in CLI? Here the output of the CLI when an external call is received: Goto (macro-vm,s-NOANSWER,1) -- Executing [[email protected]:1] Macro("SIP/Messagenet-0000003b", "get-vmcontext,200") in new stack -- Executing [[email protected]:1] Set("SIP/Messagenet-0000003b", "VMCONTEXT=default") in new stack -- Answer, the numbers are the transcoding delays.

Nick surfparadise 2016-01-16 20:29:36 UTC #6 Fix it! admin2all 2013-05-21 13:27:46 UTC #7 I went through all my configs in both the Asterisk IAX and Asterisk SIP Settings- also I checked all settings in which I can Disallow and My gut feelings says to me that it is some context issue. Nick technoboy 2016-01-22 22:08:58 UTC #8 I got this very same problem.

Elastix Elastix Instalação Problemas na configuração de trocos e ramais. (1 visualizando)(1) Visitante Favorito por: 0 TÓPICO: Problemas na configuração de trocos e ramais. But hey - that's why I am here ! I'm now following the README and am at the Enter License Key ! I solved the problem, however I don´t think it was a elegant solution, mainly because I didn´t understand what happens.