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...e asterisk box by name, then a smart DNS server can just point phones to the backup box after failure. However, since asterisk running on the backup box doesn't know about the phones, this is only half the solution ________________________________ From: Mohamed A. Gombolaty [mailto:mgombolaty@noorgroup.net] Sent: Thursday, June 30, 2005 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk failover solution Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a...

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the

Hi, I'm looking for a reliable provider to use mainly for outgoing calls in the UK, incoming isn't so much of a worry as I think I'm going to accept them over ISDN. Cheers! Steve -- Steve Foy steve@narnian.org

...implement asterisk in a department of university, but it's necessary autentication by students, login and password for example. thanks. -- Carlos Jersnimo ------------------------------ Message: 3 Date: Tue, 27 Feb 2007 14:02:44 +0200 From: "Mohamed A. Gombolaty" <mgombolaty@noorgroup.net> Subject: Re: [asterisk-users] Cisco 7960 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <45E41DE4.AC438E1C@noorgroup.net> Content-Type: text/plain; charset="us-ascii" Dear Khaled, What is the softphone u r u...

Dear All, I was trying to enable call forwarding, following the steps of the link on voip.org regarding this issue it doesn't work and the phone I am trying to implement on is still ringing. below is my conf in extensions.conf and the CLI output during the process. My configuration is : exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2}) exten => _*5X.,2,Hangup exten =>

Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and

...de my own dialplan. I don't really understand FreePBX well enough, nor do I really want to put in the effort of learning it since I can already hand-code. TrixBox just has a couple of nifty features that I enjoy to make daily life a tad easier. On 10/15/06, Mohamed A. Gombolaty <mgombolaty@noorgroup.net> wrote: Dear All, I am have experimented asterisk long before any gui was available and also currently working with trixbox, ofcourse working with asterisk directly makes you more aware but when you start deploying the system you will face management issues for asterisk, as anyone who dea...

Dear All, I am totally new in this arena and I am still waiting for my installation process on freebsd to finish, but I wanted to make sure of the following: - Call routing between IP telephones can be done regardless of who made the phones? - Asterisk does support SIP V2? - it does act as SIP Proxy and Register? -- Thx MAG -------------- next part -------------- An HTML attachment was

Dear All, I have installed Asterisk everything is OK until I tried to configure meeting room, configuration was simple enough when I try I get a message that it's not a valid meeting room, Now I don't have a Zaptel device on my machine, so I found that you will have to use ztdummy to make a dummy zaptel device on your machine and this is because of timing issues. My question is ztdummy

Dear All, I was trying to use Realtime Asterisk option for Sip users and peers + Heartbeat + Mysql Replication in order to make a failover system, so that if Ast1 went down for any reason, Ast2 server will have the same data that Ast1 used in the Mysql database and don't need to make the phones re-register. But when I started testing: the calls that where active during the transition

Dear All, I am currently working on asterisk cvs-head version in order to use realtime with mysql, 2 asterisk servers with duplicate mysql databases, one asterisk server is serving the sip phones and the data is logged to the database and replicated to the other asterisk database, when the first server fails though it has the sip phones data in it's database the sip phones need to re-register

Dear All, I am having a problem in a scenario I am doing, I have two branches, every branch has has an asterisk@home that deals with each branch locally and a trunk connected to a central asterisk, now if any branch wants to call another branch it goes from the local asterisk@ home --> to the central asterisk server and then forwarded --> to the remote asterisk@home server --> to the

Dear All, I have this problem which is preventing me from switching to voip system andstill working on that old siemens pbx, we have fax machines that we attached to ATA called planet and when we try to send a fax locally between the fax machines it works great but when we try to get a fax machine to send or recieve on the E1 pri or on a TDM400p (notice all cards are digium) it get's a

Dear All, I am facing a strange problem that I can't find any matches for while googling, sometimes while a call initiated from asterisk to the PSTN is answered and the answering party say the receiptionist tries to transfer the call to someone else, the call dies, the full log shows nothing useful and I am really unable to move forward on this issue, so can some one suggest anything? My

Dear All, I am not sure if this is the right place to ask my question but I can't find a newsgroup or support for this app_RPT concept so I hope if some one in this community who have tried it out could help me out. I studied this application requirments and saw the hardware needed they describe a radio quad which uses RJ 45 but I can't see where the RJ goes in order to be able to

Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk??

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31