search for: g726

Hi, I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of this morning 3/30/04 of asterisk, zap and libpri. The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced". When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I can see: [format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) == Registered file format g726-40, extension(s) g726-40 == Registered file format g726-32, exten...

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the SIP trace : INVITE sip:20 at 192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.16...

Hello, I'm wondering if anyone has benn able to successfully get g726-16 passthrouhg to work? I am wanting to use this codec instead of g729 as I'm running out of DSPs using a high complexity codec on the Ciscos. I would think it would work just as g729 does, which has been working fine for me, but it does not. G726-32 does work great however, but it's lik...

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered S...

With sipura (I tried this with both the 3000 and 841) set to prefer the g726-32 codec, a call from the sipura to asterisk will use g726. Asterisk sip.conf has: disallow=all allow=g726 allow=gsm allow=alaw When the call is from asterisk to the sipura, asterisk will not use g726. It ends up using alaw. I usually use stable but I tried this with head too, and same thing ha...

Hi everybody, A customer requires G726-40 with Asterisk... I know G726-32 is pseudo-standard, but he definitely wants G726-40... Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone done this before? Any hints? Please help! Due to a misunderstanding, my product manager already offered this to the customer and now i...

...vox vox slin sln sln|raw g722 g722 g722 ulaw au au alaw alaw alaw|al ulaw pcm pcm|ulaw|ul|mu ilbc iLBC ilbc h264 h264 h264 h263 h263 h263 gsm gsm gsm g729 g729 g729 g726 g726-16 g726-16 g726 g726-24 g726-24 g726 g726-32 g726-32 g726 g726-40 g726-40 g723 g723sf g723|g723sf 18 file formats registered. =========== Am I missing something in the configuration files, or maybe I'm missing some module? Thank you.

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

I am able to use sox to convert audio files from ulaw to wav (MS ADPCM), is there a way, using sox or another command line tool, to convert them to g726 ? ( g726-32 since it is supported by * ) tia, -baji. --

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday, December 06, 2006 11:16 AM To: 'asterisk-users@lists.digium.com' Subject: FW: G.726 on Asterisk 1.4.0 Importance: High This is what I found today googling...

...ers, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - - - - - - - - - - - - - - gsm - - 3001 3002 6999 3001 3000 10999 - - 40994 8000 6999 - -...

In the mailing list archives, I found a message that indicates that the IAXy has the ulaw, alaw, and g726 codecs, but I cannot find anything official on Digium's site about it. The Installation Manual has an example iax.conf file that indicates the ulaw codec, so I know that one is good. But we are thinking about using the IAXy over a VPN, to replace our MultiVoip. alaw and ulaw are 64kbps,...

...red translator 'lintoalaw' from format slin to alaw, cost 1 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1 == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 [codec_g726.so] => (ITU G.726-32kbps G726 Transcoder) == Registered translator 'g726tolin' from format g726 to slin, cost 10 == Registered translator 'lintog726' from format slin to g726, cost 10 [format_gsm.so] => (Raw GSM data) == Registered file format gsm, extension(s) gsm [fo...

Hi All, I am happy to offer $1000USD for the fix of the g726-32 in Asterisk. What's wrong with it? It currently gives a very distorted sound as though the gain is set to high. Lowering the gain on endpoints helps but this is not a fix just a poor workaround. We require g726-32 to be of the same quality as the Asterisk g711 implementation. As the develop...

...: No Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 200.93.xxx.xb Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Username : 105 Codecs : 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263) Codec Order : (g729|g723|gsm|g726|ulaw|alaw|h261|h263) Status : UNKNOWN Useragent : Full Contact : sip:105@192.168.0.253:5060 And this is the output of sip debug peer 105 when I call to *98 (for voice messages): asterisk*CLI> sip debug peer 105 SIP Debugging E...

Anyone have an example XML file for the PAP2T? Cheers, j

Hi all, I would like to make a call using the asterisk IAX with g726 - 16 bits codec. How could I configure it in the iax.conf file. Do I need to modify the file like this? . . disallow = all allow = g72616k . . I have tried it but it hasn´t worked. Thanks in advance and best regards Guild __________________________________ Do you Yahoo!? Meet the all-new...

...Sipura/SPA2000-2.0.10(e) Content-Length: 428 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 25855460 25855460 IN IP4 192.168.0.160 s=- c=IN IP4 192.168.0.160 t=0 0 m=audio 16388 RTP/AVP 2 0 4 8 18 96 97 98 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 14 headers, 19 lines Using latest...