similar to: Configuring Asterisk as Gateway SIP-H.323 via ooh323

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 =

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

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

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

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

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

[public_gk] ;exten => _070.,1,Set(CALLERID(number)=070333333${CALLERIDNUM}) exten => _070.,1,Dial(H323/${EXTEN}@59.120.139.119) exten => _070.,n,Hangup *CLI> h323 show peers Name Accountcode ip:port Formats 7000 ast_h323 203.160.252.147:1720 0x4 (ulaw) 88670333333 ast_h323 203.160.252.147:1720 0x4 (ulaw)

Dears; Actually, the needed file name to be ooh323.conf and not chan_ooh323.conf, so I copied the file from chan_ooh323.conf to a new file name ooh323.conf and it is working fine.

Has anyone tried using ooh323 in Asterisk to talk H.323 to an Avaya S8500 running Communications Manager 4 software? I have a potential customer who has such a system, and wants an Asterisk box to talk to it. Apparently they don't have SIP installed. I've successfully got ooh323 talking between two Asterisk boxes, so am just after some confidence regarding the Avaya, or any gotchas to be

Hi all, I have installed Asterisk-addons-1.4.5. I was getting error cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory So, I did following steps: cp asterisk-ooh323c/.libs/libchan_h323.1.0.1 asterisk-ooh323c/.libs/libchan_h323.so.1.0.1 make install make samples It worked properly.But still I am not getting ooh323.conf in /etc/asterisk Please help me. Am I doing

Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323 ( no matter is it asterisk with ooh323 or cisco gateway) I don't see this. Could you tell me is it possible? Thank you!

I terminate some calls on a h323 device (a quescom gsmgateway) from asterisk 1.2.3 with ooh323, the customer is complayining about choppy sound on most of the calls, the only warning message I can see is : src/chan_h323.c:944 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_102 (the calls sounds perfectly with iax/zap termination and the quescom seems to work fine with

Does ooh323 from asterisk-addons 1.2.1 support alaw codec? This is what is written in h323.conf.sample that can be found in asterisk-addons dir. The codecs to be used for all clients.Only ulaw and gsm supported as of now. Default - ulaw ONLY ulaw, gsm, g729 and g7231 supported as of now disallow=all allow=gsm allow=ulaw So, it shouldn't support alaw, but I manage to establish calls with

Dear All, It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a

Can someone tell me how I can configure ooh323.conf to accept call from h323 gateway (only the authorized h323 gateway) to my asterisk. I will be glad to know how this can be done. I tried the setting as in ooh323.conf [abcd] type=user context=default ip=62.193.1XX.2XX disallow=all allow=gsm allow=ulaw this gateway can make call, even if these lines are commented out and you restart the

anybody successfully running asterisk-callmanager scenario with h323 trunk (ooh323 channel driver in asterisk)? I'm using 1.2.12.1 & ooh323 from 1.2.4 add-ons, but seems, that ooh323 is ignoring dtmf digits from callmanager h323 trunk setup with chan_h323 is working fine with dtmf I tried all possible modes with ooh323, but without success, with chan_h323, I'm using default (rfc2833)