similar to: two sip clients on one server

Im having a trouble understanding the config files setup even with some documentation ive read such as the handbook, maybe im just illiterate. Anyway do you think some one can point me to some examples of real config files. Such as IAX, Extensions, and Sip. I just cant grasp the concept for some reason. If someone would like to help me out, maybe even explain one on one? Thanks a lot

Where can I get some asterisk books.. or tutorials..? I?ve been searching in google.. but I find just some tutorials explaining how to fast set up an asterisk server. I want to learn how to use it and how to make my own configurations. So, the thing is that I want to know what is the best book or tutorial that you know? recomendations? Thanks to everyone... Danko Miocevic

I appologize if this was already answered somwhere on http://www.voip-info.org/wiki-Asterisk, I'm sure it probably is. And if you wish to just point me to a link that would be appreciated. I am very new to asterisk and unix all around, so these questions may sound rather ignorant. First being, how do I setup asterisk to point to another asterisk server and make all the lines which should

I could use a few pointers to get this working please? I have asterisk installed on my linux server. It is setup to register with sipgate and works for incoming calls. I have xlite installed on my windows pc and this connects fine with the asterisk server and can get the incoming calls fine. Now I want to be able to make outboun calls from xlite via sipgate. I also need to be able to dial

Hi, all: I am a newbie to the asterisk and its architecture. :( After reading some help in the tarball of Asterisk, I am still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example document. Maybe what I want is too much, after all it is a open project, not commercial product. If I want to get

Hi All, I've installed digium TDM02B. The PSTN line connected to this card. At the IP network side, I have SIP phones registered sucessfully to asterisk server. How do I configure asterisk, so once there is incoming call to the TDM channel from PSTN, the caller will hear another dial tone from asterisk then to key in the extension of intended destination (SIP phones number) that already

Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer.

Hi and happy new year! My question: - two extensions: 1111 and 2222 - an active call on 1111 - incoming calls to 1111 should be forwarded to 1111 (call advice!) and 2222 I know how can I forward an incoming call to more than an extension, but I have no idea how can I get the information, that 1111 has already an active call... I think, I need something like: exten =>

Is it at all possible to set a Global Variable freely whenever a context gets used without having to enter an extension priority to use SetGlobalVar? This is really limiting the dialplan for me. Heres an example of what I would like to be able to do. [globals] AREACODE= [local] exten=_NXXXXXX,1,Dial(SIP/${AREACODE}${EXTEN}/blah) [anyoldcontext1] AREACODE=313 include=local [anyoldcontext2]

Basically I am trying to make it so I can dial an extension and it will pick up an fxs line and bridge me to it. It's to integrate it into an old intercom style system. So basically there is no ringing.. I dial the extension and it picks up the line and we are instantly connected. Any ideas? Thanks! -Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

as i am trying to use asterisk and install my newly purchased ( got it yesterday) digium cards. i am following the very detail steps of http://www.automated.it/guidetoasterisk.htm. but one thing did not seems right so i wanted to let enveyone know the page says: Once compiled make sure there is a copy in /usr/bin/mpg123 i think the location is /usr/local/bin/mpg123

Dear asterisk users, We want setup TE121 digium board: Model: Digium TE121: VoiceBus technology allows the TE121 to use an industry standard bus-mastering PCI Express interface. http://www.digium.com/en/products/digital/te121.php My platform Server: HP Proliant 150 G5 OS: UBUNTU x86_64 GNU/Linux Asterisk: 1.4.21.2 zaptel: SVN-branch-1.4-r4662M When we enable zaptel driver for TE121, the

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

I'm experiencing some problem using the transfer() application,expecially when a call in received from a queue. I'm using Asterisk 1.4.22.1 This is my scenario: ; this is the piece of code in extensions.conf that place the call in the queue when 1111 is called exten => 1111,1,Answer exten => 1111,n,Queue(2000|t) ;this is the piece of code that calls the user test when 2222 is

hey all, I'm using openssh-3.5p1, was trying to set up a 'reverse telnet' session (sun solaris 2.6 on both machines). Anyways, I was doing: server% ssh -R 1111:<server>:2222 <client> client% ssh -p 1111 <client> where <server> is behind a firewall and <client> cannot reach <server> Anyways the idea was to connect to the socket on

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm very new to Asterisk and I have the following scenario. 1. Let's say I have a number of 1-222-222-2222 from my SIP service provider (VoicePulse). 2. I point my phone, Verizon wireless cellphone (1-111-111-1111), voicemail to the number provided by SIP service provider (1-222-222-2222). 3. I use another phone (1-333-333-333) to call 1-111-111-1111 and leave a voicemail message.

Hi .. I've just received in the post my Wildcard card with a single FXO and three FXS daughter cards. I've identified a dedicated PC to function as the * machine and installed the card. I've installed Fedora Core 2 on that machine. I've downloaded the * software and the zaptel drivers. And now, to be quite honest, I haven't got much of a clue what to do next! I've

Hello, I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc and I got this warning (this was also before I

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