similar to: ulaw to g726 conversion

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) ==

Just a quick listing of tested, and updated, steps from my notes. Enjoy ! http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html -baji. --

hello, running asterisk 1.4.11 on CentOS 4.5 I am getting no response on function STRPTIME() the system just hangs, STRFTIME() is working fine as seen below. Same thing happens whether I called in from a softphone or via teliax. While executing the following code : ; exten => s,n,Set(v_ts=) exten => s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)}) exten =>

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 like Asterisk doesn't

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

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,

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

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

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

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!

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

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"

Dear Folks, I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is simple: connection to the PSTN directly via SIP, using g729 codec, and connection to the softphones (X-lite 3.0 build 56125) trought local network, using ulaw codec. Sometimes, I got messages like: [Jul 1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported SDP media type in offer: image

Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says "Request sent." The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but

Hello list, I have always though codec translation table is dircetly connected to system speed, utill i came across this: in my lab, i have 2 boxes, First box is an Intel Celeron 1.7 GHZ with 256M RAM: show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw

about something else than the IP Trunk, he is talking about outbound (which is related to using an application to run an outside call, which is used usually in campaign in contact centers and so on), I think nthis case differs that placing a calls via IP Trunk or even outside call but the caller who will do it (and not the application). Lastly, Mr. Amit helped me when he gave me a configuration

May be slightly off topic, but I was wondering what everyone thinks of this latest ruling against Vonage? Does anyone really know what Verizon hold patents for, and could those patents possible affect anything in Asterisk? Who knows who Verizon will go after next. Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I'm trying to build an Asterisk packages for a C3 system (256MB memory, cpuinfo below). /proc/cpuinfo: processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 9 model name : VIA Nehemiah stepping : 8 cpu MHz : 1000.736 cache size : 64 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu

I'll take either.... Actually now that I have had a chance to think about what I did (sorry bad week here). Yes, I will admit I did patrionize the users list... sorry if I offended anyone. I just figured I'd try to help any SunRocket users out that may not be on the biz list. If you review my history, you'll see I only post business stuff to the biz list. This is an exception.