similar to: GnuGK + SIP Provider + Asterisk

Hi guys, I create a topology like fellow: /****** /************ /*********** * GK *-----------* Asterisk *---------- Sip Prov * ******/ ************/ ***********/ | | | | | | H.323

Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with

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

Hi! I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323 built with needed PWlib v.1.5.2 and open H.323 v.1.12.2. But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's compiling fails and I get error 1. Do you have any working solutions with asterisk and

Hi, I'm using asterisk with chan_h323 together with gnugk. chan_h323 and gnugk were recently compiled with pwlib-1.5.2 and openh323-1.12.2 as advised. When connecting asterisk directly by ohphone (without gatekeeper), everthing is fine. When using gnugk for usage control in routed mode, I find a funny situation in asterisk's H.323 debug: == New H.323 Connection created. --

Greetings- As most of you who monitor this list know, I've been messing about with Asterisk -HEAD, Cisco Callmanager, and the nufone H323 channel driver here for some time- with pretty decent success. I'm hoping to cash in a chip here- I've run into something that is probably a very simple answer, yet not found a decent reference to resolve it. Scenario- -HEAD as of last week

Current setup: Polycom IP3000 <-> gnugk <-> asterisk <-> Cisco 7940 Asterisk and gnugk are on 10.20.98.6 IP3000 is H.323, using G.711 (10.20.98.2) 7940 is SIP, using g711ulaw (10.20.98.3) I've been asterisk for a while now, only using SIP devices. I'm happy with that side of things, but I've not used H.323 before this week, in trying to get the IP3000 to work. *

Hello folks. We are trying to interconnect an asterisk installation with a gnugk 2.0.5 installation to become able to use some H323 hardware that needs a gatekeeper (particulary an Ericsson WebSwitch 100). We have managed asterisk to dial H323 endpoints successfully (although calls are interrupted immediately after connection with "spawn extension exited non-zero"), but we could not

Hi, I also had some problems using chan_oh323 together with gnugk. * <-> gnugk <-> h323-phone When I called the phone and hang up, befor the phone was picked up, the h323-phone continued ringing. The same, when the h323- and some sip-phones were called, and the sip-phone picked up the call first. (It is annoying, when you are talking to someone at the phone and the phone on the

Hi, i'm trying to setup Asterisk to work with GnuGK using the Openh323 channel driver. I have a Gatekeeper that gets H.323 calls from a Cisco GW. To this Gatekeeper I've registered some endpoints, Cisco ATA186, Snom 100, etc. Now i want send the numbers 083xxx into Asterisk. Easy, i'll just enter something like this into oh323.conf: gwprefix=083 And all my calls starting with 083

Hello guys, I am here again, sorry for borring but I in freeze here! : ) Just for resuming some doubts: + The extension.conf file: has a [general] context for "general" configurations, and a [global] context for global variables. The another context [any_name] are context for "handle the calls", for example: exten => 1001,1,Dial(SIP/1001@10.11.2.121,30) means, when the

Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h323.conf like: [office] type=h323 prefix=9 context=outbound I get a message saying:

Hi I've a problem with a gnugkv2.0.7 I've compiled gnugk successfully I've installed PWlib-1.6.6 and openh323-1.13.5 libraries successfully When i run gnugk i have this error: error while loading shared libraries liboh323_linux_x86_r.so.1.13.5 cannot open shared object file No such file or directory I try to use command export: export

hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &

Hello, I am trying to configure asterisk with GnuGK. Is it possible to configure Asterisk OH323, CHAN_H323 without register to GnuGK gatekeeper? thanks, Ganbold -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060118/5f99c433/attachment.htm

Hi All; A2Billing is working fine for Asterisk, but in case I need to use Asterisk and Gnugk and I need to manage the accounts and the billing from one Database and one billing system, so I need a prepaid billing that can work with both. Which prepaid billing (open source) can be used to work with Asterisk and Gnugk? Regards Bilal

Hello, I need to install a h323 channel in order to asterisk act as a sip/h323 translator . I want to use gnugk in full proxy mode for the h323 terminals nated . Which h323 channel for asterisk and gnugk h323 oh323 or ooh323c ? Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver

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

Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, I have a client that wants me to replace their existing H323 gateway. I am able to get ooh323 and h323 to work fine in a native environment, but the whole thing goes to heck when I have to cross networks. Gnugk seems to be the answer to this, but I can't seem to get it to work right. Any ideas? Thanks Danny Nicholas