similar to: Can Asterisk hear on two IP addresses?

Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ------------ ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

Hi List; Is there a link that help me to configure asterisk to register to another gatekeeper as client? Regards Bilal ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

Dear Alex; Thanks for your kindly help and answer. The question here is: how asterisk will be able to receive calls at two network cards where each network card has a different IP address. Maybe we need to know if asterisk is doing a hear on the ports only without caring for IP or it is doing a hear only on the IP:port? Any advise? Bilal, There is no technical difference, from Asterisk's

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Hello, I'm looking to do the following, and I wonder if Asterisk can be used for it, and if yes, if anyone can point me to the relevant information (commands, sample config...): 1. Caller dials 111, 222 or 333. 2. Based on the dialed number, Asterisk queries an external MySQL table and retrieves alphanumeric data, plays/announces it to the user and deletes the row from the database: The SQL

FLAC 1.2.0 is out. There are a few new features and some speedups and fixes, but more importantly, there are some small changes to the decoder to pave the way for possible future compression improvements, so applications developers are encouraged to upgrade (the API has not changed). The changelog has all the details, but in summary: - automatic SSE OS detection at runtime (so no need to

Hi List; Can asterisk hear (receive) calls on two IP addresses? How? If yes, then: If I have a VPN router, and my Asterisk server connected to two network cards, one has a private IP address (192.168.0.2) connected to the VPN router (192.168.0.1) and another network card has a private IP address (193.111.196.249) connected directly to the outside default gateway (193.111.196.240), where the VPN

I'm a research student and I need to modify openSSH in a way that I can implement my algorithm in it. My algorithm mostly involves buffering the packets which are received by the user using some formula to remove the timing information in packets. Also the second part is to encapsulate SSH packets into RTP packets while communicating with a modified SSH server. The client embeds SSH packets as

2007/7/25, Josh Coalson <xflac@yahoo.com>: > > FLAC 1.2.0 is out. There are a few new features and some speedups and > fixes, but more importantly, there are some small changes to the > decoder to pave the way for possible future compression improvements, > so applications developers are encouraged to upgrade (the API has not > changed). The changelog has all the details,

Hi All; Anyone knows an IAX IP Phone works fine and tested? Does polycom support IAX IP Phone? Regards Bilal

Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination

I'm using asterisk 1.4.5 , on Centos. My kernel version is 2.6.9-55.ELsmp I've configured the nway call. I made entries in extension.conf, feature.conf, as per required. I'm trying to make a 3-way conference with the 1 user myself ( using asterisk), and two others are PSTN line users. I'm making a first call , then putting that person on hold by pressing **( as per feature.conf

Dear all I have video confranceing deivice Codian and i want to intergrate asterisk box with codian so voice confrance is possible with codian users means some users have not codian endpoint so thay call join confranceing with SIP PHONE I have configures asterisk and register codian in asterisk now whn i call from asterisk to codian i got IVR and ask me to inter confrance

Hello; I need to use Cisco IP Phones with Asterisk and I have some questions to know how to use it if someone can advise: 1) How I can assign for each button an extension? 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? 3) As you know that it is required to have a correct username and password to login, so where to give the username and

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Dears; To patch libpri: I just place the patch file in the libpri source directory and then I run make and make install? Or I need to compile the dahdi and asterisk also? If the problem stayed, do I have to go for previous libpri version? Or for previous dahdi version and asterisk version? Regards Bilal ----------- > bilal ghayyad wrote: > > But I am afraid it is a bug because I

I'm trying to use menu.c32 to boot different kickstart files. The problem I have is regardless of what menu item I select, it always uses the client.cfg (default entry) even if I select one of the others. I have removed the DEFAULT from that entry and it still always uses the ckient.cfg kickstart file Any ideas. Thanks, Mike Jarka default menu.c32 prompt 0 timeout 300 MENU TITLE PXE Boot

Hi Danny; Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager? Regards Bilal ------------------------- It depends on how you are configured. The gui interfaces using Asterisk Manager, so you get the Same IO from the gui that you would get from a native manager session. -----Original Message----- From: asterisk-users-bounces at

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk