similar to: GRQ

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'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

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:

Hi, I am working on a H.323 project which involves GnuGk and Asterisk My current goal is to provide IVR functionality for the H.323 users which register through GnuGk(eg. call credit information) I have successfully built a H.323 platform using GnuGk - it uses SQL accounting and authorisation. Now I am trying to integrate it with Asterisk in order to provide IVR functionality as I already

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 have built latest Asterisk w/ OpenH323 channel driver. We have a SIP softphone registered to the Asterisk. We can place outbound calls from the SIP phone to the PSTN via OpenH323 connection to our gatekeeper. Everything works okay - DTMF and Audio... But in the reverse - if we call from a cellphone or landline the PSTN number we can get the SIP phone to ring - we answer and can hear the

Due to the format of the message coming from the H323 channels included w/ Asterisk we were unable to use our gatekeeper. For a quick solution we tried the OH323 channel drivers and can receive inbound calls from the parent gatekeeper. We are trying to do a dial to gatekeeper... I am trying exten => 5551212,1,Wait,2 exten => 5551212,2,Dial,OH323/5551212 But I am not sure if this is the

Take mercy on me - I'm a newbie w/ Asterisks... Here's what I'm trying to do - and please someone let me know if this can be done... We have a commercial VoIP network (we are a communications carrier)... The gatekeeper (Lucent iMerge) supports MGCP/H.323 and allows for calls to be made to the PSTN cloud via GR303 links. I would like to build Asterisks with H323 (or MGCP if need be -

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.conf, followed by the relevant portion of the h323_log: [general] port = 1720 bindaddr =

Hi there, I am trying to register asterisk to a Lucent's MVAM GK. It is not registering to the GK with both h323 channels(chan_h323 and oh323). The problem is that if I set the GK(in asterisk) through the GK's IP, the GK answers with GCF without it's GK ID, and after that it does not answer to the RRQ, because the RRQ message has no GK ID. And when I set the GK through

I've complied the OH323 .so successfully and can easily receive calls from my H323 gatekeeper (using 711u), however it seems that all outgoing calls are refused and I'm getting "reason 23 (Temporary failure)" as an error code which I can't find documented everywhere. My H323 gatekeeper needs a 001NXXNXXXXXX to dial out to the PSTN even if I'm in north america (Montreal)

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

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:

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

For the benefit of everyone, having H323 Configuration problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems

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

I've got problem with connections h323 -> sip and sip -> h323. I've Cisco 7940 phone with sip soft and Netmeeting as h323 node. As gatekeeper I've gnugk and brand new asterisk from cvs + chan_oh323 0.5.5 When I call from Cisco (SIP) to h323 node by alias registered on gatekeeper and h323 node will answer the phone... I have on my Cisco still Ringing. Call termination, no

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

Hi there ! I searched the whole web to find some helping information about H323 Control Protocol, but there is no way to find that information. We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2 + 'asterisk-oh323_1.5 channel driver + wrapper' and configured the dialplan for using our H323 Endpoints which are ip200 Innovaphones. Besides, we also use Gnomemeeting but

Actually I am facing a problem with H.323 (the standard and the ooh323) with Asterisk vesion 1.4.25 and I discover the following: 1) Using the standard h323 that come with Asterisk: The chan_h323.so it is not existed in the /usr/lib/asterisk/modules after doing the compilation and installation for (pwlib, openh323, /chanels/h323, asterisk), although make menuselect was done and the h323 channel