similar to: iLBC, Speex and X-Lite

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

Hello, I know that my Kphone question may be a bit off topic, but I have been busy with this again and again for about one month now, sent three mails to kphone@wirlab.net (the contact address mentioned on http://www.wirlab.net/kphone/index.html), asked for a solution in a german ip phone forum and tryed many things by myself. I try to compile KPhone 4.0.3 (tryed CVS Version as well) but

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

Hi, Has anyone managed to get X-Lite to work with Asterisk using the iLBC codec.. I have just tried updating the the latest version 1079 (BTW this new version supports up to 10 proxy configurations, Not that I can see a reason to have 10 proxies setup, I would rather have the ability to transfer calls).. I can make a call using iLBC but the sound that I hear is just a lot of pop's and

Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

Hi all, as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions: moloch*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202

Hi kind developers, I need select soon the best freeware VOIP codec, I see that all competitors are using iLBC because of the separate packets management. How speex behave in case of packets drop? Why other choice all iLBC? Thank you for any kind answer. Best regards. ------------------------------------- Roberto Della Pasqua Http: www.dellapasqua.com Email/Msn: roberto@dellapasqua.com

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

I am using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57. I have tested both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

Dear Asterisk experts, I've just downloaded Asterisk CVS version (since I'd like to manage its configuration from RealTime). Next, I have kphone on the same Linux host, and VmWare virtual machine with Windows and X-Lite IP phone inside. I successfully tested the demo's with X-Lite, but failed to hear something with kphone (v4.1.1). There were NO problem with this kphone and stable

Make sure you are using G.711a, G.711u or GSM codecs.. I have not been able to get iLBC to work and someone the other days couuld not get SPX working.. You will need to enable/disable the codecs in X-Lite.. If you also want to control the codecs that * uses then put the following in the general section of your sip.conf disallow=all allow=alaw allow=ulaw allow=gsm Hope that helps.. > Hi,

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Hello all, I've started implementing iLBC on some of the ATAs we have floating around clients' homes, but I'm coming against this error message with most of them: codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)? The ATAs in question are various Grandstream models - the HT486 being the predominant one. Looking at the list archives, it's

hallo all, could sombody please help me, i dont know why nativ bridging is not working when i choose the ilbc codec, with speex it is working,?? iaxcomm (ilbc) ---> asterisk --> ( asterisk2 --> sip grandstream (alaw) ) \-----------------native bridge------------------/ 1. if i use on iaxcomm as default speex, nativ bridging between iaxcomm and my sip phone is working 2.

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom