similar to: Transferring H.323 Call

On our tests using asterisk, some calls have been terminated abruptely with exact 185 seconds. This is happening with all our incoming calls from a trunk from 1 of my DID providers ( other providers or trunks are fine) and I could reproduce it by calling a queue from my Wengophone Softphone and letting the MoH play for 185 secs. If I make the same call from my WRTP54G on the same place,

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

hi, i'm just look trhough the todo file, but it's rather large. i assume only a small part of it will be finished for 1.0. what is the schedule for 1.0? what kind of features will be included in 1.0? what is in 1.1 etc.. it'd be useful to see in the todo file that what features scheduled for which version. yours. -- Levente "Si vis pacem para

Hello all, After many time reading a lot of stuff I am quite confident using LARTC to route my trafic. I am still working on QoS (by package type and so on) but it will stay in my studing class for a long time... ;) So lets go to my question... I mounted a router that makes my conections throug 2 external interfaces. Its working fine and my default gateway for entire network behind

Dear all, We are experiencing some trouble when compiling R packages using R 2.6.0 and the new Rtools installer under Windows XP. (1) First, compiling any package using the new setup stops with an errorrelated to some "/" issue on the inst folder. This folder only contains a CITATION file ---------- Making package FLCore ------------ adding build stamp to DESCRIPTION installing

hi all I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h software can only be configured to point to a gatekeeper. I know I don't need to do this, but it's a test before I setup my Symbol Netvision phones (I don't have an access point for them now). First, this is what I have, and what

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and

Hi, I am investigating an issue with voicemail and realtime. What we are seeing is the following: 1. Caller calls in and goes to an IVR 2. Presses 101 to go to voicemail 3. app_voicemail start and tries to connect to the database trhough res_config_mysql. However, it takes too long to be able to connect (~15 minutes) It seems like it first attemots to connect to the database on 16:25:03 and

Hi, I'm thinking of setting up Asterisk for H.323 video phone clients. Now, what is the difference between native H.323 that come with Asterisk and "Open H.323 for Asterisk" ? TIA Kuni -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter mailto:kuni@ej-interpreter.net Macintosh Webcast Specialist

when i try to install openh323gk i get the following error please help ldaplink.cxx:345: call of overloaded `pow(double, unsigned int&)' is ambiguous /usr/include/bits/mathcalls.h:143: candidates are: double pow(double, double) /usr/include/g++/cmath:427: long double std::pow(long double, int) /usr/include/g++/cmath:423: float std::pow(float, int)

Hi Everyone, I was hoping someone might know why I am experiencing a problem with Asterisk logging the event: [May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission 03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com for seqno 669371069 (Critical Response) This is happening after: - call is setup, 2 way audio - call can function correctly for up to 5

Hi All, I need to setup the following rights behavior trhough samba and I'm currently stuck after lots of unsuccessful tests. Maybe one of you has an idea or a solution to this problem... Here it comes: - A share must be available only to some users belonging to the "project" group. That's easy: valid users = @project - There are several administrator-created

Hi, We've been working a lot with Asterisk in SIP for over 6 months but I've finally succumb to the pressure of H.323. I need to find a way to do what we do with SIP but with H.323. That is to have calls from H.323 peers placed into their own unique context (unique to the endpoint placing the call into Asterisk) within Asterisk so this is obviously done using REGISTER's within SIP but

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

I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration

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