similar to: Asterisk and Tandberg Gatekeeper

I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN). When chan_ooh323 first loads it tries to establish a connection with the gk but I it fails. I have the following extract from the ooh323 log. Can anyone give some insight? Thanks! MD 23:02:59:045 Sent GRQ message 23:02:59:045 GkClient Received RAS Message 23:02:59:045 Received RAS Message = { 23:02:59:045

All, I'm running (attempting to) ooh323 with Asterisk and a Cisco 2621XM router operating as a H.323 GateKeeper, however when I bring the Asterisk box up it registers successfully with the GateKeeper (exchanges GRQ/GCF, then RRQ/RCF) it notes the GateKeeper supports keepalive at 300 seconds, when it gets to time to re-register its sends an RRQ again and gets rejected with RRJ (unspecified

Hi, I'm trying to configure Asterisk as SIP-H.323 Gateway via ooh323, but I have an error relatively to the GK Confirmation message. >From the log: "H323 RAS channel creation - succesful Sent GRQ message Gatekeeper Confirmed (GCF) message received ERROR:No Gatekeeper ID present in received GKconfirmed message Ignoring message and will retransmit GRQ after timeout Error: Failed to

Hi All, We have a Tandberg VCS System for Video conferencing and a customer running AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing. Problem Statement: How do we integrate the 2 systems such that Audio SIP calls are seamlessly passed between the two. Sorry we're just starting up so a bit of general advice, or a link to any document would be great! If anybody has done this -

All - NEED HELP BADLY.. Using Asterisk with H323 Channels.. Everything is installed but we cannot register w/ Lucent gatekeeper. We ran ethereal and found that it was making GRQ (Gatekeeper discovery requests).. We had provided the name of the Gatekeeper (it's IP) and cannot determine why it's trying to do a GRQ. We want it to go straight into RRQ or ARQ and skip the GRQ.. Netmeeting does

All, I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working. I have the following: [Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323] 192.168.1.5 192.168.1.6 192.168.1.7 the Asterisk box has numbers

I posted previously that my Asterisk is failing to register with parent H323 gatekeeper. It appears that it is issuing a GRQ <GatekeeperDiscoveryRequest> even though I have the file <h323.conf> saying exact IP address of gatekeeper... No discovery. I'm trying to just bypass - remove the GRQ and have it Register <RRQ> Does anyone know if asterisk-oh323 will work better or

Greetings to All, I hope someone has already gotten this working. I spent all day today trying to get ooh323 and gnugk to run on the same box. After a lot of tweaking to get everything compiled, I got both up and running. I can make calls IAX to H323, but cannot make calls in the reverse direction. I have tried many different configs on the GK, but always come up with the same error. It appears

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

Hello again. I'm stll struggling trying to terminate 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

Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti ; ooh323c driver configuration ; ; [general]

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

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

Hello, I've been trying to send traffic to a Cisco Call Manager 3.2, but with no luck. Here's whats happening: * Call gets to CCM * Call gets to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk:

Tag '0.5.2' created by Benjamin Otte <otte at gnome.org> at 2007-08-24 15:25 -0700 release 0.5.2 ("Something you can do with your finger") -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQBGzup1vMv5VCdLq8QRAkWeAJ9SXSMd1qhpy/ojTxWJk2wIeBB1xgCfYA2X grQ/UmvMw8GySgs/GlnKEzQ= =r+xH -----END PGP SIGNATURE----- Changes since 0.5.1: Benjamin Otte: back

On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote: > Hi! > I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide > a conference bridge for an existing Avaya PBX. I have no control over the > Avaya system, but I am able to speak with the admin in charge when I need > stuff done. I am running all this in a VirtualBox

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

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 All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's