similar to: Get rid of H323 problems for 100$

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

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

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

Hello, ? I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we

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

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Hi all, Sorry if this has been answered previously, but I have not had any luck trying to find it. I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2, kernel 2.6.8-1.521) to connect to a gateway that can only support H323. I have installed the asterisk-oh323 channel driver (version 0.6.3b) using Open H323 1.13.5 (patched as per asterisk-oh323's instructions) and PWLIB

Hello, Could somebody tell me is it possible to use asterisk without RTP proxying in SIP<->H323 calls? I mean exactly what canreinvite=yes option do in SIP<->SIP calls. I don't need a transcoding, only a signaling conversion, and this is possible with some softswitches, so i wondering what about asterisk. Same question about H323<->H323 calls I'm using NuFone

Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hello, I have installed oh323 channel driver. Outgoing calls to H.323 world do not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that userInputMode=RFC2833 has already been set. Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel? Kind regards, Fernando Herrera _____ De: Fernando Herrera [mailto:fherrera@iplan.com.ar] Enviado el:

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

Hello, all How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf [writesound] exten => s,1, Answer exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729) I'am using oh323 channel driver, in oh323.conf

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

Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =

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:

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

Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So,

Hi all, I have seen many posts on the Shorewalllists dealing with H323. Although lots of them indicated that this is difficult process with kernelrecompilation etc. I just tried what seemed to be logical for me. Surprisingly it worked. Configuration: WS1 ----- FW ------ Internet ------- WS2/Shorewall WS1, FW and WS2 run Redhat9 with its standardkernel 2.4.20. FW and WS2 run Shorewall

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -