similar to: G726-16 passthrough...

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) ==

Hello, I'm a new Asterisk user and I hope I haven't missed something, but I can't seem to find an answer to this issue. I have a Cisco SIP gateway terminating calls into a 7960 phone. The issue I would like to fix is if I have an incoming call without an ANI, such as directly from my TDM phone switch, Asterisk says the call is coming from the IP address of the Cisco gateway,

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 do not know how to do it... Thanks a lot in advance,

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 happens. Anybody know how

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

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. --

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 developer who fixes this issue you will

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 My Yahoo! - Try it today!

-----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

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.

>So in my sip.conf I put many variations of what I thought should go in >there, >finally including (to no avail): I have had to play this guessing game as well with other codecs/config file settings in general. And I think this email touches on something that has been troubling me for sometime. >You should allow "g726" only. I think one problem area of documentation

Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth > core show translation paths alaw --- Translation paths SRC Codec "alaw"

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

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 =======================================

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

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Hi Users, 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

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

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

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