similar to: WG: G729a error

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

Firstly, I'm running Asterisk 1.4.4 on Solaris 10. I have several different internal SIP phones all sharing a single IAX2 VoIP channel. PHONES |------------- <SIP/uLAW> --------------| ASTERISK |-------------- <IAX2/g729> ------------|VoIP/ISP The g729 codec has been registered successfully and appears to be detected by Asterisk (NOTE: I have changed what I thought might have

Hello, I wonder if anyone else has noticed this. I see a pair of calls to pipe() within the codec_g729a, and suddenly, I have a leaked file descriptor that remains until asterisk dies. Now, maybe no-one sees this, mainly because I have no g729 licenses on the machines where this happens. And conversely, I haven't yet studied servers that do have licenses. Why have codec_g729a.so loaded if

Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be

Hello Everyone, I just upgraded to the latest beta version and I am running into one problem. We purchased g729a licenses from digium and they aren't loading anymore. If I roll back asterisk to 1.2.10 the codecs work fine. I've downloaded the new 1.4 version of the codec from their website and re-registerd everything with no luck. Here is the error message: error loading module

2009/5/8 Olivier <oza-4h07 at myamail.com> > Hello, > > Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are > instructions to install G729 software. > (I think I followed instructions step by step but g729 license doesn't seem > to show up). > > My question is : > Is the command bellow still up to date ? > > >g729 show I suddenly

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

Morning all, I updated to 1.4 now but it seems the oracle is not working with it? I get error with 1.2 all is fine: Mar 29 08:10:54 WARNING[3876] config.c: Realtime mapping for 'sippeers' found to engine 'oracle', but the engine is not available Mar 29 08:10:54 NOTICE[3876] chan_sip.c: Registration from 'sip:xxx@xx.xxx-xxx.de' failed for xx.xx.xx.x- Username/auth name

Hi, I've been trying to get the G.729a beta codec running with my remote Asterisk box that talks IAX2 to my local Asterisk box. Digium fixed the problem I was having in registering the beta codec, so that now works fine. I've removed the old codec_g729b.so from /usr/lib/asterisk/modules and put in place the codec_g729a.so beta from digium FTP. My CVS build of Asterisk is about a

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

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

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, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile:

Hi, I have a Cisco 5300 which I want to make a call THROUGH the Asterisk PBX (security) to an IP phone which supports g729, and vice versa. Both Cisco and the phone talk this codec if I do not force the call to go through * However if I say canreinvite=no in the sip.conf for either of these gadgets, the call will fail with No compatible codecs! I have bought a 5 user license just to

Hi, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

Hello.... I have the follow situatuion: < ISDN > | | V E100P |----------------| IAX2 / g729A |----------------| T100P | Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - - -> |--------------| | | | | | Zhone | ----------------- ----------------- --------------- Here's the situation: I receive calls from the PSTN

Hello! I am wondering how the differences between G729, G729a, and G729b effect call bridging and server interoperability. For example, can one server use the G729 code with another server that uses the G729A codec? Also, which version is Asterisk set up to use? Thanks! Elliot

Hello, When I'm trying to play standard sound files from Asterisk using G729A codec with OH323 channel I get this message: channel.c:1650 ast_set_write_format: Unable to find a path from GSM to G729A It seems that this files must be in G729 format? How can I convert this files to G729? ... or am I wrong? -- wbr, Oleg

Hi all, Will G729A codec exhaust the CPU power? If yes, how many concurrent sessions that P4 server board that can stand? Pls advise. Btw, if G729A has been purchased and installed, what will happen to the Asterisk Server crash say hard-disk when down or faulty, any where to do back up first such as "tar" commands? Any advice will be appreciated tq

Hi, We need help in enabling g729a codec for our SIP peer that's using Cisco AS5300. Our codec is purchased from Digium. We are able to dial out the numbers and answer the call, but there's no audio. This is when only g729a is allowed. We noticed when they also allow ulaw codec on their side, the codec used falls back to ulaw and the problem is gone. -------------- next part