similar to: Oh323, Please Help Newbie ;(

hello, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call:

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?

Hello- In order to satisfy a customer requirement, I've just build H.323 under asterisk (using the specified versions of OpenH323 & PWLib, and trying to follow the instructions religiously), and it seems to have come up fine. When testing with with OpenPhone (Windows version 1.8.1) and NetMeeting, I've gotten some intermittent results however. All my calls are from a PC to asterisk -

I figure that I've successfully loaded and compiled the h323 module into asterisk I can successfully place a call from openphone to a sip phone (snom200) So I figure that the h323 module is working. The question I have is how do I return a call from the sip phone to openphone? I get an error message Sep 7 17:09:49 NOTICE[110992304]: chan_h323.c:861 oh323_request: Asked to get a

Would someone kindly share some definition into the following? Openphone version 1.91 includes dual sets of Speex codec's starting with: SpeexNarrow-5.95k{sw} SpeexNarrow-5.95k{Xiph} Through SpeexNarrow-18.2k{sw} SpeexNarrow-18.2k{Xiph} I do not understand what the differences are between {sw} & {Xiph} given the same bit rate for both? Are all of these Narrow or Wide or Ultrawide

I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we

Hello all, I just installed Asterisk with H323 support (chan_h323 from Jeremy McNamara). But experience problem while connecting OpenPhone to Asterisk Here is h.323 trace: 5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720, handle=27 5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225

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 --

My apologies if this question has been answered previously. However, I found that it was nearly impossible to search and find since anything can cause a segmentation fault. Problem. When Asterisk is booting up the h323 modules and a client tries to connect before Asterisk/h323 is finished booting, the program seg faults out and doesn't load. I thought about putting this into the inittab,

hi all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Can somebody help me? Thanks, Thomas.

Hi, I run asterisk with oh323 plugins.It runs correctly with sjphone H323 Gatekeeper. But When i run openphone it doesn't recognize my asterisk server like a gatekeeper !! What is the problem ? Thx

Hello I am a newbie to Asterisk. We have set up Asterisk on a PC with Redhat 9.0. We have installed H323 openphone on our PC's. We are wondering what a gatekeeper does. It seems we need one but what I have seen in this group is that a gatekeeper must be installed on another box on the network. As all our PC's on the network use Microsoft OS is there a free gatekeeper software for

Hi there. I am trying to connect Asterisk to a local danish ip-telephony provider. But is having some difficulties. First I thougt they were related to the provider. But then i started debugging on the Asterisk (aix2 debug) When I make a call using AIX to the provider everything seems to work just fine: *CLI> -- Accepting AUTHENTICATED call from 192.168.1.150, requested format = 1024,

Hello again. I'm stll struggling trying to terminate calls from SIP through Asterisk and throught my H323 gateways... Basically the call is accepted by GnuGK but then dropped with *reason = unreachableDestination <<null>>* I did a *debug trc 10* on GnuGK and looked at the sessions... one from X-Lite through Asterisk... and one from OpenPhone... The one from OpenPhone works

> http://www.speakfreely.org/ > > I think this would be one of the best real-world tests of the speex codec. > This software doesnt use ACM or directsound api's but uses straight C code. > I was thinking the speexenc/speexdec should be easy enough to add. The last time I looked at this it was still very much old news - mostly half duplex audio, does not adhere to any

HI all, Any of you having experience with voice master? I tried using the openh323 channel it doesn't give me voice at all. THere's no packet coming in. There's no problem with any other equipment but voicemaster doesn't send voice at all. Funny thing, i have an old version of OpenPhone, it's working. So please if any of you knows this problem, please share. THx a bunch

hello i was searching for solution to problem (sip->h.323). any one from this list asterisk mailing have any idea how to fix it. i am getting error when i try to call from sip to h.323 user i am successfully registering my asterisk box with gnugk. but when i try to call to h.323 openphone on working on GnuGatekeeper, asterisk is not routing it to GnuGk. i am getting the following error. do

Update on this problem: I gave up on the "native" h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I