similar to: Ring an incoming call in multiple extensions

Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then continue. Asterisk console says: analog: -- Attempting call on Zap/2/3391818250 for 104@inbound_originate:1 (Retry 1) > Channel

Hi asterisk GURUs I have an issue with a call forwarding setup, I have the next zapata conf: -----------------zapata.conf----------------- [channels] context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 channel=1 -----------------zapata.conf----------------- And in extensions.cof I have a

Can any body give me an example how to configure h323 in Asterisk. Which files do I need to configure? just extensions.conf and h323.conf ? Thanks, Patricio _________________________________________________________________ Descubre la descarga digital con MSN Music. M?s de un mill?n de canciones. http://music.msn.es/

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]:

Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone "Allow Auto Answer by Call-Info: yes Dialplan: exten => 501,1,SIPAddHeader(Call-Info: answer-after=2) exten => 501,n,Page(SIP/140&SIP/110,d) exten

Hello List- I'm having a problem getting snom 190 phones to transfer a call to another local extension. Here is the scenario: A call (call1) comes in from the PSTN to (exten1). (via pri, if that matters) Another call (call2) comes in to (exten1). (call1) is put on hold while (call2) is answered. (call2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Hello, I'm running into an issue with TLS transport and I am probably missing something obvious. We are trying to configure an extension to use TLS for the transport. The extension can make outbound calls using TLS, but inbound calls fail. The extension configuration in sip.conf is set to transport=tls. To keep things simple for testing, the dialplan looks like: [default] exten =>

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

Hi, * gurus, I wonder if there is a way to alert the in-house extension(s) in case of an incoming external call without actually answering it, before somebody picks up the phone on one of the extensions? This way the caller wouldn't have to pay for the call until somebody answers. Regards, Christian Lademann

I am setting up * to accept incoming calls and route them to our reps. What I'd like to do route the call to the rep who has been idle the most, thus distributing the load among the reps. I can't seem to find this functionality. Can someone point me in the right direction? Script Head

I have a TDM400P with 2 FXO cards and I'm using Asterisk@Home 2.8 I noticed that when I place a call to the analog lines from outside, Asterisk takes a while to actually ring the extension the call is being sen to. I've been doing some tests, calling from my cellphone and here is what I see... - After the first ring on my cell, Asterisk logs to the CLI that is has an incoming call -

I just got a TE110P up on an XO PRI - everything looks good so far. We've been given a block of 23 numbers for the PRI. If I explictly set the incoming extension in extensions.conf like: exten => 1153,1,Answer or: exten => _XXXX,1,Answer I can get the incoming call. If I try and do: exten => s,1,Answer I'll see something like this: -- Extension '1153' in context

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >