similar to: ooh323c h323_convertAsteriskCapToH323Cap Don't know how to deal with mode 0x40 (slin)

Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")

I have searched and searched for over a week on this but can't seem to find the issue. Calls from CallManager to Asterisk are being disconnected immediately. I have setup CallManager and Asterisk per Shaun Ewing's pdf http://asterisk.edropbox.net/ccmasteriskvm.pdf I have installed Asterisk 1.4.0-beta3 on Fedora Core 5. I got libpri, zaptel, and asterisk compiled and installed.

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from

Call-Bilal*CLI> module load chan_ooh323.so Loaded chan_ooh323.so [Jun 17 20:23:32] NOTICE[2392]: chan_ooh323.c:2506 reload_config: Unable to load config ooh323.conf, OOH323 disabled Loaded chan_ooh323.so => (Objective Systems H323 Channel) Again, from make menuselect, if I selected chan_ooh323 from the Add-ons and I selected ADDON from module embedding. Then I ran ./configure and make. I

Hi, I am using asterisk 1.4. I have confgured the musiconhold.conf file. However, when i make a call and then hold the call it does nothing. in the CLI i do not see the starting/stopping musiconhold messages. i am making calls from sip to h323 using asterisk assip/h323 gateway (with gnugk and ooh323). i get the following messages when putting the call on hold: -- Executing [204 at default:1]

I try to get H323 to run, but have so far only partial success: There is a Gatekeeper GK, where asterisk connects to. The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper. From the Network on the GK, asterisk is reachable via the number 070333333. I have an extension on asterisk 6002, which is reachable. I try to call a number attached to the gatekeeper (070168177) with the

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 -

Hello, I have asterisk 1.4 with ooh323c addons installed. (As I am a newbie in voip world...my question might be idiot...! ;) Please forgive me!) I succeed to make H323 call when ooh323c is configured as gateway (gatekeeper=DISABLE in ooh323.conf). When I put gatekeeper= ip_address, and add an account as follow : [aaa] type=friend username=aaa password=xxxx host=dynamic context=test

Hi, I have two asterisk servers where softphone A is connected to asterisk A. On those two asterisk servers, ooh323c is installed. I tried to call a "test" context on asterisk B from softphone A. But I always fall into context "default" of asterisk B. ( I don't know how to tell asterisk A extensions.conf to call asterisk B "test" context) Here are conf

hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk

I''m newbie Xen-4.1.1 I''m trying to boot DomU with the following configuration. Sometime DomU boot, but sometime boot failed I''m not sure what is the problem. I can attach that image with tap-ctl on Dom0. Does anybody advice me about this problem?? Sincerely * Environment OS : Scientific Linux 6.1 x86_64 Kernel : 2.6.32.43