similar to: # Transfer context problem

Hello all, I got the following error compiling h323 support in the latest cvs. Below the error is a diff to the file that I got to make it work. I took an example out of sip as far as the syntax for ast_rtp_new. Not sure if it is correct or not, but it seems to work. Please correct me if I am wrong in the additional 2 arguements. Regards, Scott cc -g -pg -c -o chan_h323.o -march=i686

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. --

I've been banging my head on this one for a few days and am quite stuck. I've got a gatekeeper running and everything works there. Netmeeting works calling other netmeeting clients. Netmeeting calling asterisk connects, but netmeeting can't generate the signals to make the demo do anything other than talk. But connection from ohphone always disconnects straight away. I can't seem

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 friends Can you please say me "How to send an extension number from ohphone to astersisk". For eg I have an extension 5454 at the asterisk. How can I make a call to that extension from ohphone. I tried with the command ohphone 5454@IPAddressOfAsterisk. But I could n't call that number. I want to do it without using any gatekeeper. Can you please suggest me the solution?

Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

Hello I'm using H323 channel and client used ohPhone-1.4.1 (with gatekeeper). when at client side dial to asterisk server (dial 7777, test mode). ohPhone don't hear any thing sounds (no audio). i dial between ohphone (with gatekeeper). sounds are good. my current setting. Asterisk-1.1.x, GNUGK 2.2, PWLIB-1.8.3, OpenH323-1.15.2, ohPhone (for windows) 1.4.1 Please help me. Thanks

Hello all, I tried to compile Asterisk from CVS yesterday. I would like to try gnophone as IAX client on the same Asterisk server with a sound card. I didn't have any problem of compilation. Here are the config files in /etc/asterisk: /etc/asterisk/asterisk.conf /etc/asterisk/extensions.conf /etc/asterisk/iax.conf /etc/asterisk/indications.conf /etc/asterisk/logger.conf

I'v got the following scenario: Two clients (ohphone) are calling (one at a time) the host with asterisk, which then connects to the SIP client. One of these hosts let's asterisk crash with a segmentation fault (i can provide the core file, if needed) in the second, the SIP client accepts the call. However .. if that client get's to the voicemail instead, because the SIP client is

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi all, I'm using asterisk in a production environment now and this afternoon I got reports complaining that it was not working. Looking at the asterisk console output, I saw it contains lots of error messages as printed below. Unfortunately it is not obvious from the logs as to what started all this. Just before the error messages start, everything seems to be working fine with no problems.

Hi, all Please help me. My platform is RedHat Linux 9.0. I have a wildcard x100p. I just installed asterisk by following step: # cd ../zaptel # make clean ; make install # cd ../libpri # make clean ; make install # cd ../asterisk # make clean ; make install # make samples When I test Asterisk typing # asterisk –vvvvc I get one error and one warning: [chan_iax.so] => (Inter Asterisk

Hello-- Why do I target chan_sip for so much effort? Because, it seems to me, chan_sip is probably the most used channel driver in the asterisk community!! (and, of course, the zap/dahdi driver, is also pretty popular) I haven't had time to follow up on chan_sip, and I probably won't for several months. But, if I had time, here is what I'd do: There are two ways to speed up

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

Joost Witteveen (joost@iliana.nl) wrote: > > So, each UDP package with 20 bytes speex-data, we send: > > 20 bytes speex > 12 bytes ogg headers (and others?) > 28 bytes UDP/IP headers (2 IP numerbers, 2 portnumbers, checksum, etc, etc) > > and, if it goes over the phone, each package has a few ppp headers. > > Am I overlooking something, or does this fixed frame

I've a problem with my X100P card. I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I call an PSTN phone number, some digits are duplicated, so I'm unable to call the right person. Not very clear ? I'll try to do better (sorry, I'm french...) example : I use ohphone (with quicknet hardware), I call asterisk (*192*168*1*204#), asterisk answers, I choose

Hi! When I try to make a call with ohphone, that is the message I get: Call to "06302" aborted, insufficient bandwidth Can anybody tell me a solution or a reason why this messages appears? Thanks a lot! Regards, Mireia

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered