similar to: DTMF Echo on H323 Channel

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like also to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

Hello all, I just installed chan_h323 by Jeremy (nufone) with asterisk-head. well, I have some questions if someone please briefly answer. 1 - can we configure h323 to work in g729 pass-thru mode, like we do in SIP. becaz I had to installed g729+intel libraries to work with g729. where as we don't need to install g729 to work in pass-thru mode. 2 - now most important question is

I'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

Hello everybody, when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286 and H323 EP (my post from 13/08/04) I checked further and discover that problem is with ATA286 who is unable to call. I always get an 404 error. Coming back to RC1 everything works fine again. I tried to modify my dtmfmode from rfc2833 to info but in change nothing. Local call to asterisk are

Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might

Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No

Hello, I need to install a h323 channel in order to asterisk act as a sip/h323 translator . I want to use gnugk in full proxy mode for the h323 terminals nated . Which h323 channel for asterisk and gnugk h323 oh323 or ooh323c ? Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver

Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2

Hi, I'm trying to compile h323 channel driver on cvs Asterisk 1.0 but no success (I get a lot of errors - related to pwlib library). I read in docs that there is also 3rd party h323 channel driver (somehow both even share protion of code?). I wonder what are pros and cons of both drivers ? Should I try to compile native driver ? Thanks in advance, Robert.

I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the "Up" state, with asterisk consuming 100% of CPU: *CLI> show channels Channel (Context Extension Pri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None) (None) 1 active channel(s) *CLI>

hello, sometimes per day, below messages appears in my asterisk/messages log... any suggestion, what this mean? thx PJ Mar 20 07:57:53 WARNING[4672] channel.c: Prodding channel 'H323/ip$172.20.1.11:53473/331' failed Mar 20 07:57:53 NOTICE[14280] chan_h323.c: Avoiding H.323 destory deadlock on ip$172.20.1.11:53473/331 Mar 20 07:57:53 NOTICE[14280] chan_h323.c: Avoiding H.323 destory

Hi group! I'm having problems with ooh323 (ver 0.3?!? - the one that comes with asterisk addons 1.2.1) and I need to know what h323 channel driver you use in production? Have a nice day! -- Tomislav Parcina tparcina#lama.hr

I am having the issue described in this question: http://lists.digium.com/pipermail/asterisk-users/2005-May/099075.html Does anybody has an insight? I guess Asterisk is trying to match the combination IP:Port, but in H223 this changes call by call. There is no way to add "insecure=port" like in channel_sip.

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hi, I'm on a RH 9 box, fresh install. When I try to compile channel h323, I get multiple compile errors. Can someone help? asterisk, ptlib, openh323 all are fresh from CVS. Thanks! Here's what I've done so far: All the source is in /usr/src/asterisk/ [root@localhost openh323]# ls -la /usr/src/asterisk/ total 164 drwxr-xr-x 8 root root 4096 Sep 25 00:59 .

I have following settings done on my Fedora8: Downloaded openh323-v1_19_0_1-src-tar.gz pwlib-v1_11_1-src.tar.gz Extracted them in /root/openh323 and /root/pwlib Exported the following variables: PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH Then I compiled pwlib and it was fine. But in

Just starting my search for a SIP/MGCP/H323 channel bank. I need analog ports in a building that only have network connections back to my * server. I could install another * server and use a normal CB with a PRI but I would like to investigate any good CB's with network trunk abilities Thanks Dave Packham

Hello, I try to place a call between CISCO IP phone 7905 (h323) and an analogue telephone on DLINK DG-104S (mgcp). Call setup is handled ok - both way, but after connection of the call I get both site no-audio, even codec are Set to alow-64 both. The call is terminated with this debug output - Didn't get a frame from channel: MGCP/aaln/1@dlink-1 and is Hanged up. Jan 18 20:32:55