similar to: SIP to H323 translator

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards

Dear Alex; Thanks for your kindly reply. Please explain for me what do u mean exactly in "a la" in the following sentence u wrote it below? " in SIP, this can be done via "re-INVITEs" a la the canreinvite= option for SIP peers in sip.conf" Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full

Hi All; How can I configure Asterisk in that way: If I entered code (from my mobile when I call to the Asterisk or from any Internal Phone), then the machine do restart. I need this when I am far from the office and I need to restart the machine and I do not have Internet connection. Any help? Regards Bilal

Dear List; To have better security, how can I put a password on the international calls (if the user dialed the international call, then it will be asked for password to send the call outside)? Can this password read from the CDR file to know whom did these international calls (using which password? As I might have mutliple passwords). Regards Bilal

Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hi All; In dahdi: what we use instead of ztcfg -vv (that is existed /sbin/ztcfg -vv). ? Regards Bilal

Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels

Hi All; Anyone tried to install Asterisk based on UNIX (not linux)? Which UNIX was good to work with Asterisk? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

Hi List; I am trying to use wiki via the link (http://www.voip-info.org/wiki/index.php?page=Asterisk) in effective way to find the needed resource for me, but still it is hard to arrive for the needed information. For example: what is the best (shortest) way to search for information related to the command playbak()? Using the backlines, it make the eyes feel hard by keep reading without

Hi All; When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem? Regards Bilal

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to

Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk to register firstly then I can route my calls to that SIP trunk. In IAX2, we use the register => , so what shall we do in Asterisk? And how its format will be (if we will use register)? Or what is the solution? Regards Bilal

Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I

Hi all; I have two asterisk boxes installed in two separated sites, the internet bandwidth between them is very good and I am using G729 codec to communicate with them and IAX. The problem that side A hears well side B while side B does not hear well side A !! I did one thing in side B that in iax.conf, I set the bandwidth=high and it helped, but still side B is complaining from the quality

Hi All; It look like there are some free (open source) tools that are used for Asterisk reporting special for call center (to see number of agents logged in, number of calls now, .. etc), and to be used as dashboard. Can someone direct me for something really is suitable and stable? Regards Bilal

Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -------------- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad

Dears; In asterisk 1.8, it is not more possible to use DeadAGI? Also, I found the below commands in the a2billing and I would to ask why it set the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? How? [a2billing-callingcard] exten => _X.,1,NoOp(A2Billing Start) exten => _X.,n,Answer() exten => _X.,n,Wait(2) exten => _X.,n,DeadAgi(a2billing.php,1) exten =>

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

Hi List; Can someone advise me if Polycom support H323 that work fine with Asterisk? And wether this H323 Polcyom devices more costly than SIP Polycom. Also, I am not able to know if new Polycom come with PoE adaptor so no need for PoE Switch (can use normal switch that does not support PoE)? Do I need any special cable for Polycom or normal Ethernet cable? Regards Bilal