similar to: Re: [asterisk-dev] Dynamically Adding A Context

Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi all, I am using Asterisk CVS, and I am getting a huge delay in dialing SIP. This Asterisk box is taking calls from a PABX over ZAP, then dialing SIP users. So, a user '0251' dials from his phone, the PABX sends it the my Asterisk box, no delay, then I get a 15 sec delay, before it actually dials the end SIP user. 1 -- Accepting call from '0251' to '0834541083' on

Hello List, I am getting Asterisk set up. I am going to be installing Debian Linux on a laptop later. I would appreciate some detailed instructions on: (A) What to type into the shell to download and install Asterisk. (B) How to open the configuration files (*.conf) (C) If there is a way that I can change the configuration files remotely (SSH?). Thanks in advance. -- -Shane Blog:

Hi, I've looked at the Wiki for this, have seen the Swift.agi details, but has anyone got a current script for Cepstral and an example of integraton in * please? I'm a * and linux newbie, so please be gentle ;-) Thanks John

Greetings! I've searched far and wide for an answer and have gotten no where, so I was hoping one of you guys might have the answer; Is it possible to dynamically add a context to the dialplan? You can add extensions via the CLI, however if the context doesn't exist I get an error message instead of it creating the context for me. Any method will do, AGI, AMI, CLI... I just need a

Hi, I just want out find out how to do bill recon's when you send calls to a provider. They send me their CDR's, and when I compare it to my * CDR's, some calls are 1 second off, either way. How in general is it done by others? -- thanks, Yusuf

Hey all, I have read on voip-info.org that to configure MoH asterisk requires the use of mpg123. I have installed mpg123 and restarted asterisk. But, when i put a call on hold i get this error: May 25 14:13:03 WARNING[1872]: res_musiconhold.c:865 local_ast_moh_start: No class: default Can you help, Thanks yusuf

Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw

Hi, I have a bunch of SIP phones(behind NAT) registering on my * box. I want to find out when they register and de-register. I also want to operate on it, so when they register/de-register, I want to insert calldate into a mysql DB, etc..... Maybe this will help me when, for instance a user tries to register but has the wrong username/password. Now I am aware of regcontext, but it only

Hi i am planning to develop a php script that will be called from AGI for the management of an IVR application. I'd like to be able to do the following things from php: - retrive callerid - play some audio files to the caller - wait for some DTMF digits - retrive the DTMF - stop the call the php have to collect some information from the user and after some check on a database inster some

Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker "SIP peer authentication on an external database (RADIUS - LDAP)", etc. Although, none of these seems to give me the

Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 <- 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]:

List, I finally decided to break down & start playing with AGI scripts, but for the life of me, I can't figure out what I am doing wrong. Below is a super simple script to run a query in mysql to see how many call records there are for the extension calling in, then print the total in the CLI. This is all I get on the CLI: -- Executing AGI("SIP/216-0baa",

I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I

Hi, I have xlite client registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==>1500,1,AGI("localhost//hello.agi") So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard Thanks &

After testing some AGI's, I noticed several extra Asterisk processes. They are not zombies, but can't be killed by safe_asterisk. Nor will they die when CLI issues stop now. Then I read that each AGI spawns a separate Asterisk process. But all my AGI calls have apparently completed successfully. So there should be no reason for them to hang there. Several questions: 1) Under

Hello, I'd like to run Asterisk on an embedded device, where space is scarce. It should be able to handle calls from a VoIP provider in SIP, calls from the PSTN through Dahdi, and voicemail. If someone's already done this, I'd like to know which directories/files are required for a basic install? Does this look right? ================= /bin/asterisk /etc/asterisk/ asterisk.conf

Uuugh..for the life of me, i cannot delete sound files using "EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)" through AGI the AGI debug log indicates the command executes successful ( equals 0) but my files are clearly still there. If i try System(rm ...) in my extensions.conf diaplan it'll work there. Is there a bug in the AGI to use "System" ? because i tried to

Hello I run AGI scripts from extensions.conf to save data into an SQLite database file, but this file must also be accessible in read-write mode by PHP scripts served by Lighttpd. As far as I can tell, Asterisk runs by default as root:wheel. I don't know if AGI scripts also run as root:wheel. Lighttpd runs as www:www, and if I create a new SQLite database through PHP scripts, they're

I'm presently working on a project to build a scheduling system accessible by both web and phone. on the web side one can query what items are available when by using the time or the item as a key then reserve for an available time slot. reservations may also be modified by the user that made them or an admin. Where may I find examples of doing similar things with asterisk? all I've