similar to: chan_h323, Asterisk and DTMF issue

How do I get asterisk to populate the "Calling Party Number" field in an H.323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the "Display" field rather than the "Calling Party Number" field. -----Original Message----- From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com]

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what

I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [asterisk@jonux h323]# make clean rm -f *.o *.so core.* [root@jonux h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN

I'm trying to use H323 for the first time so please forgive me if I've made a mistake here. I have downloaded and compiled the latest versions of pwlib, openh323 and asterisk. I have dtmfmode=inband in h323.conf, but the remote system is not hearing the DTMF. Running a trace reveals the following... 1:08.398 ThreadID=0x00022012 h323.cxx(4594) H323

hi .. I have finally managed to get Chan_H323 & G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib & oH323 with versions taken from nufone's site

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

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 --

I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make

Hi, Someone released a succefull interconnection in H323 with WTL equipement ? I'm trying to do that with an IPNx. But get dead air. With chan_oh323 it's fine, all works. With chan_h323 => dead air. The configuration is GW to GW. This is my configuration from h323.conf: [general] port=1720 bindaddr=my.ipaddr dtmfmode=rfc2833

It appears Asterisk can handle DTMF inband on only a limited selection of formats, of which G.729 is not one. The issue appears to be something involving "short data" -- whatever that is. (I'm inferring all this from looking at dsp.c in the vicinity of the error message I was getting, which pointed to line 1424.) What *is* "short data"? Is this really a show-stopper for

Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can "hear" me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external

Hi all, I try to ask helps again about chan_h323 extensions. I define this in h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 allow=gsm allow=ulaw gatekeeper = DISABLE context=default [gm1] type=friend host=192.168.1.20 context=default [gm2] type=friend host=192.168.1.25 context=default and I have in extensions.conf : [demo]

Hi, Can anyone confirm that if I want to do h323 proxying that I do not need codecs installed? For example if the codec being used is g723.1, I don't need the codec installed locally because there is no compression or decompression being done on my server; the incoming traffic is simply being sent out on another h323 channel (h323 in->h323 out). Is this correct? Thanks, Chetan

Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

Hi All; I am trying now to compile h323 to be able to use it, I did the pwlib and openh323 successfully and I exported the PWLIBDIR=/usr/src/pwlib_v1_10_0 and the OPENH323DIR=/usr/src/openh323_v1_18_0, then I was need to compile h323 as following: cd /usr/src/asterisk-1.4/channels/h323 When I type make, it gives me: make: Nothing to be done for 'degault' And when I type make opt, it

Just thought I'd share my success with chan_h323 and our Symbol NetVision phone (4046-100-US). Voice quality is excellent, and setup was trivial. The new NetVision firmware (4.21) is much better than the 3.x stuff. It gives the phone a whole new look and feel. The hardest (and longest) part was getting OpenH323 compiled. After that, H.323 ran out of the box. I simply uncommented