similar to: list of pbx control sequences?

Hi, I compiled asterisk and chan_h323 on an Opteron in 64 bit mode. In the h323's Makefile I replaced in line 24 CFLAGS += -march=$(shell uname -m) by CFLAGS += -march=k8 and also tried CFLAGS += -m64 -march=k8 Both solutions do compile, but when starting asterisk, a load error occurs: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi When I grep

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. --

IPManager version 1.6 has just been released. Below is a list of some of the features you will get on your Asterisk server using IPManager to generate your configuration files. Download: http://ipsoftware.thorben.dk <http://ipsoftware.thorben.dk/> PBX Features The following features will be available to users of the PBX if you are using IPManager to configure your PBX. *

hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know whether this's * or Snomes' prob. Any help would be appreciated.

Hi, how can I disable event_log in order to reduce hard disk activity? I can't find any hints in conf files. Must I hack the source code or even use brutal methods like creating a dir called event_log in the log dir, in order to prevent asterisk from creating an event_log file? (Just chmod a-w event_log does not work, unfortunately.) Thanks for any hints! Roger.

Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to

Hi, a partner, who exchanges voip traffic with my asterisk box, complains, that asterisk ignores hints about ports to use. Hints about ports to use, seem to be a feature of H323. (I'm not firm enough with H323 to verify this.) The remote party opens the media-in channel: remote-ip:port-A -> local-ip:port-B My local Asterisk-box uses the same channel for media-out: local-ip:port-B ->

Hi, I don't know quite how to ask this question, because my knowledge is so limited at this time. I have an h323 phone that I am trying to use to do VOIP to phones on the PSTN. I want to sign up for a service and not have it go out my POTS line. I do have a Quicknet Line jack in my RH 9 box and it is fully confiugred. I have downloaded the latest drive from openh323.org and installed it

Hi, I have unpleasent short audio gaps when a perl based agi scripts starts. Thus, I now started to put all those things in C programmed daemons for fast-agi. Anyway I'm looking for another mean, which would help me more quickly. I noticed, that all agi scripts are running with system priority -11, like asterisk does. This is really waste of priority. I would like to have the AGI scripts

Remote Call Pick up feature is very much implemented in asterisk. You can pick up a call for another extension by dialing *8# To be able to do that, you need to have the extensions in the same pickup group, configurable through sip.conf and zapata.conf. -- sudhir > ------------------------------ > > Message: 14 > Date: Fri, 27 Aug 2004 14:17:26 +0000 (UTC) > From:

Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323. Cisco conf: dial-peer voice 8 voip destination-pattern 2... session target ipv4:<asterisk ip> codec g711alaw no vad h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ;tos=lowdelay ; disallow=all allow=alaw allow=ulaw allow=gsm context=from-internal extension.conf [from-internal] exten =>

Hello, I am looking for a way to control another legacy PBX from Asterisk using a CTI interface. Are you aware of any legacy PBX CTI control card that can be controlled by Asterisk? I have an Avaya PBX with CTI interface and researching if I can connect Asterisk to this. :-) Thanks for any hints. -- - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/

Hi list, After several tries to understand the subtil description in the h323.conf to be able to make the next scenario I was presented the following error messages by asterisk. Can somebody tell me please what I am doing wrong. Scenario: Gatekeeper (h323) --> Asterisk PBX -->(h323) Gateway Endpoints are connected to Gatekeeper. Call does come in like 999931235650087 with codec g711

Hi, some days ago we discused here the need for an analyzer for the 1000 Hz tone, as opposite application to Milliwatt. Here it is: Mwanalyze http://planinternet.net/download/voip/asterisk/app_mwanalyze.c It performs a Fourier analysis for a fixed frequency and tells the amplitude. The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple

Hello! Tried to compile, but : [CC] chan_h323.c -> chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing pointer to incomplete type chan_h323.c:350: error: dereferencing pointer to incomplete type chan_h323.c: In function 'oh323_rtp_read': chan_h323.c:790: error: dereferencing pointer to incomplete type chan_h323.c:791: error:

Hello all, I have problem. I can't makel asterisk addon, asterisk-ooh323. I use Asterisk and addons svn version. OS:redhat EL4 Linux 2.6.9-5.EL #1 Wed Jan 5 19:22:18 EST 2005 i686 i686 i386 GNU/Linux Please help me . [root@asterisk asterisk-ooh323c]# make make all-am make[1]: Entering directory `/usr/local/src/asterisk-addons/asterisk-ooh323c' source='src/chan_h323.c'

I have a SIP phone (Cisco 7960) registered to 2 * pbx, is there anyway to control which * pbx will be used for making calls? I know by default the cisco will use and I want to change that. thanks micko

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the "Up" state, with asterisk consuming 100% of CPU: *CLI> show channels Channel (Context Extension Pri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None) (None) 1 active channel(s) *CLI>

Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find