similar to: h323 still no success to dial out via GK

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

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, 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, 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

Hi. It looks like the gpg signatures of the RC2 and RC3 tarballs are bogus... I've tested the .bz2 and .tar.gz files downloaded from the us1 and us4 mirrors. To confirm that it's not a local problem, I've tested the signature of the Samba_CA.crt file (http://us4.samba.org/samba/ftp/) and it's ok. Can someone else please confirm it? Thanks. - Ademar $ gpg --verify

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

Hello group, I am trying to register to a opengk h323 gatekeeper using chan_h323. The gatekeeper expects me to register a username like 31201234567@gatekeper.com with a password secret and an e164 of 31201234567. Thus I put the following in the config file: [general] gatekeeper=w.x.y.z. AllowGKRouted=yes [31201234567@gatekeer.com.com] type=h323 e164=31201234567 secret=geheim

hello all, i have a conceptual question. i have a h323 gateway and it is connected to a h323 gatekeeper. my question is: can i connect my gateway to another gateway directly? i mean can these two gateways work with each other without working with gatekeeper? or when i have connection with a gatekeeper, all calls must be set through it? thanks in advance, SAM -------------- next part

Can anybody spot the problem? H.323 via GK calls Asterisk box and should be connected directly to the extension 6002 (an voice sip phone) The caller hears only ringing. The called party hears the ring. Called party picks up, caller hears still the ring tone. Called party hears nothing. Called party hangs up, caller hears the busy tone. extensions.conf: exten => 88670333333,1,Wait(1) exten

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 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]

Hi all, I am trying to get Capi Dial to use a specific outgoing msn. I can't get it to work. If I make a test call to 0703241494 (same isdn line, just one of the other numbers) I don't get CLID at all. Any ideas? ; use 0703241432 as outgoing msn exten => _070.,1,Dial(CAPI/@0703241432:${EXTEN}|30|r) in capi.conf I have: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8

hi, hi we have G729 codec from Digium, without the G729 codec, we can do the hash transfers to other sip phones fine. but once we are using the G729 codec, the asterisk is not responding to hash transfer, ie, when we press "#" it does not detect it and says "transfer..", is this a problem with G729 codec? (for testing purposes we have bought licenses for 2 chs) this also

Hello ! I am experiencing some problems with Asterisk trunking, this is the scenario: There are 3 servers, a DID server provider (VOIP provider) which delegates us a bunch of DID numbers to our asterisk server number one (I will call it AA), from which I route the calls to Asterisk server number 2 (I will call it BB), which then terminate on phone handsets. The trouble is, that I probably