similar to: H323 to SIP

Hi all, I'm new to asterisk and had just install it on my linux server. Can anybody told me how to setup it up for interworking with cisco h323 voip gateway? I check throught the manual on http://www.digium.com/downloads/marketing/asterisk.pdf but cannot find any information for configuring h323. Cheers. Raymond Lau -------------- next part -------------- An HTML attachment was

Apologies if there is a cleanly written and searchable FAQ that I could be directed to. I have no problem to RTFM if I can find the FM... Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... BTW, where would I find a useful FM? David -- David J. Sussman, MBA email:

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

Hello! I see many of you experience troubles with H323 stack. I am focusing on building H323-SIP Asterisk based softswitch with all codecs supported (including G729 and G723). I can setup Asterisk from scratch with H323 support or solve your h323 nightmare with existing asterisk system for for 100$. Contact me pls offline.

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 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

Hi, I am using the asterisk-oh323 wrapper and I am looking to allow registration of h323 endpoints and allow Asterisk to act as a gateway. The idea is simple: H323 endpoints would register with Asterisk. They each would have their own internal extension (like SIP). If a H323 endpoint dials an outbound extension, then the h323 call gets routed to a H323 Gatekeeper which then terminates

Hello, can i use different audio codecs when i calling between sip devices ( snom phones ) and different when i making call from isdn to sip or from sip to isdn ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A

Hi, I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! All I would appreciate if someone could advice me on how stable is sip-h323 & h323-sip translation as well as how many calls can it handle when doing such translation.( assuming single 2.8Ghz intel processor & 1GB RAM) Regards, John -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

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Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

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,

During debugging of H323 trunk side (using Jeremy Macnamara's H323 driver in ~/channels/h323) a couple of things come don't quite work as advertised... 1/ the following line in extensions.conf explicitly sets the outgoing caller ID (required in my case for downstream GK processing..) exten => _1NX.,1,SetCallerID,6400047602100 exten => _1NX.,2,Dial,H323/${EXTEN:1} what

Hi everyone, Could someone help me on make my Asterisk registers to a Gatekeeper. I have compiled the chan_h323.so and it seems to be working. What I want to know is how can I "route" my SIP clients to a single account on a remote Gatekeeper. I have tried a lot of conbinations but nothing happend. For example: my account number: 123456789 ;extension.conf exten =>

Hi All; First of all, I am looking to have the H323 Gatekeeper service available at Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing H323 gatekeeper functionality or not? Until 1.4.26.2 version, there is no h323 gatekeeper functionality. So, any implementation for this feature has been done in the other versions?

I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4) The pwlib + opal packages don't satisfy Asterisk's configure script (to let H323 compile), so I removed those and added the latest ptlib + h323plus (from h323plus.org) I can compile ptlib and h323, but when I load chan_h323 in asterisk I get a segfault. I had to point LD_LIBRARY_PATH to /usr/local/lib with the

Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am trying to use native h323 built from asterisk 1.2.7. I configured the h323 to receive incoming calls...the problem is i can receive the call to my asterisk and it rings another extension but no audio. I don't see any good documentation about gatekeepers, fast start, etc with h323. I would like to get some help from you guys to fix this issue. If any of you have configured

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