similar to: Channel h323 and oh323 fails to match inbound IP

Hello list, I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider I have this problem: oh323 (last version): ------------- asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable

Hello list, I have many IP hardphones like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider ( G729 from digium ) I have this problems: oh323 (last version): ------------- asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider (

Hi, what's the difference between h323 and oh323 modules? which one should I use? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030722/3d3edb73/attachment.htm

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

What is the difference between H323 and OH323 in Asterisk? I need Asterisk to have basic H.323 support so we can offer some simple H323 termination for some of our Cisco and Quintim hardware. Our upstream provider uses SIP, so I figured I'd use Asterisk as the go-between. I already setup Asterisk so it can push calls out through our providers via SIP. I just need a good/solid/very simple H323

I m new to asterisk n i've got an IP phone that supports h323 protocol.... but i dont know how to configure asterisk to use it... i m comfortable in using sip & iax softphones.... but there is no h323.conf in /etc/asterisk/ .... i read that i've to compile some files but i m confused regarding h323 & oh323 ...... which one should i use.. plz tell me or atleast give some helpful

i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Debian has a package for chan_oh323 (the original, external h323). It is not maintaind for quite some time AFAIK and also AFAIK offers no real atvantages over chan_h323. So I'd like to remove it. Before I do that, I have some questions, as I'm not familiar with H.323 channels: 1. Are there any useful features oh323 supports that h323 doesn't? That the version of h323 in 1.4.21

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I've just finished my upgrade to asterisk RC2. I need to have H323 support, and in the last months i've been using the chan-oh323 with good results. My question is: anyone in the list have made tests with both chans (oh323 and h323), which is best ? For this installation i don't need the gatekeeper support, i just want to

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

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.

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through

A very strange problem.. * dies with seg fault when calling from SIP to h323 WARNING[589848]: File chan_oh323.c, Line 2429 (alerted_h323_connection): Call with reference 183 in unexpected state (4). -- Called 113506 -- H323:183 answered SIP/alex-1e48 Segmentation fault I'm using Mandrake 9.1 , lastest * ( 3 day old from cvs) , and oh323 last version.. here are my configs:

Hi all, I got serious problem here, I hope I ask on the right place here (sorry if I am wrong). I have used asterisk 1.2.17 with openh323 ver. 0.7.3, for integrating between SIP Gateway and H323 Gateway, it runs about 6 months. But, recently I think it doesn't work anymore...I can't call from SIP Gateway to H323 Gateway. I try to debug oh323 by using : # oh323 debug toggle But I got no

Hi! I want to make a call to/from Panasonic IP pbx thru asterisk via H323. H323 is working nice, I can call/receive using netmeeting -- Executing Dial("OH323/WindowsUser@195.184.x.x-305e28cf", "OH323/11@80.98.x.x|15|tr") in new stack -- H.323 call to 11@80.98.188.159 with codec(s) alaw -- Outbound H.323 call to destination '11@80.98.x.x', channel

Hi all (Michael), how it is possible to get the ip address of the calling party ? (i know by using h323... but there're a few unknown segfaults...) and so i want to use oh323, but i have to get the ip from the caller to permit or deny the call with AGI. Is it possible at all ? Thanks, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger

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

According to the wiki at voip-info.org, the dial structure for using oh323 without a gatekeeper is: OH323/<exten>@<host>:<port> or OH323/<exten> The second option is valid only in the case where a gatekeeper is used. NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the destination host. When this version is used then the above syntax should be: