search for: openphone

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 don't have an outbound requirement. If anyone has successfully made either of these combo's work, could you please suggest some area where I may have gone...

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 Nar...

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 channel of unsupported format '...

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 set codecs allowed in the following order: G729 G...

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no h245Tunn...

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 Started incoming call thread 5:37.4...

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 -- Called...

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 microsoft operating sys...

...in new stack -- Called demo@demo/6000 -- Call accepted by 212.130.58.212 (format ILBC) -- Format for call is ILBC -- IAX2[demo]/5 answered IAX2[demo@192.168.1.150:4569]/4 -- Attempting native bridge of IAX2[demo@192.168.1.150:4569]/4 and IAX2[demo]/5 But when I am trying using OpenPhone witj H.323 it seems like something is not working: *CLI> iax2 debug IAX2 Debugging Enabled *CLI> -- Executing Dial("H323:10", "IAX2/demo@demo/s|40|r") in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00001ms SCall: 00003 DCal...

...te 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 fine. And i found something on the beginning of the session... Here's the session from OpenPhone: 2004/09/21 14:52:35.219 2 RasSrv.cxx(2224) GK Read from 216.119.135.3:2367 2004/09/21 14:52:35.223 3 RasSrv.cxx(2237) GK admissio...

...ere to any signalling standards at all, and not being currently maintained. LinPhone (www.linphone.org) using the OpenSIP stack has used Speex from day one, and it's in wide use. Very real world. Speex is *already* supported in the OpenH323 stack (www.openh323.org) and in use in the ohphone, openphone, Gnomemeeting, etc. VoIP implementations that use that library stack. Very real world. :) Greg <p><p><p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'sp...

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 you have any idea. please help i am stuck here for a week. i am unable to find anything on google on this topic. -- Executing Dial("SIP/2000-ae3f", "OH323/4050@gnugkip:1720&q...

...e 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 NetMeeting (for the first time). Notes to others who want to try OH323: * The installation is a bit more complicated than h323. Follow the instructions in the ReadMe file exactly. * You must choose and install the proper versions of PWLib and OpenH323, as stated. * Don't forget to...

...windows for production but I need something to test on right now. Not having too good of luck with anything. estara crashes during registration. Looks like it keeps trying to register for some reason and the client crashes. x-lite won't even attempt to register, it just sits there. I tried openphone but the codec is messed up -- just garbage but hey dtmf works :P The linux box that asterisk is running on is the only extra one I have right now to use for linux. Kphone was the only client I could get compiled. Won't exactly work running on the same machine. So if you have any other ideas...

...t (IP) | Linux PC with * and AVM Fritz! ISDN Adapter | ISDN | Someone with a analog/digital phone (POTS) Basically, people sitting on their PCs will wear a headset, and whenever they want to call someone, they start a phone application (e.g. Openphone) and dial the external/internal number. This software contacts *, and * establishes the connection (notifying the local/LAN user, or making a call through the ISDN interface to the external number). Additionally, incoming calls to the * gw are routed to the LAN PC where the user with the corres...