similar to: Dialout handler with/without leading 1

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

I currently have two fax machines on my system. Both of them seem to send and receive very slowly. My end users are complaining; saying it was faster before we moved to * (Straight Analog Lines) Any help would be great. PS: I already have the d option on the Dial line. Both fax machines are in their own context: [faxes] exten => _9NXXXXXX,1,StripMSD,1 exten =>

hi, i installed asterisk and works very well, the only problem is that when i try to call a direct number of a company that has a normal PBX i got this error: to 10.8.210.153:5060 == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro) -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack -- Goto (doisdn,00115601992,1) --

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

I need to be able to dial a long distance provider that uses an access number and an auth code. I would like to be able to program this so that the user can dial 8 and then the long distance number, asterisk will hopefully do everything in the middle. The sequence to accessing the provider is on my traditional phone speed dial as: * Dial local access number * Wait 5 seconds * Dial the auth

For years we've had the following simple context for outgoing calls: [outtrunk] ; match any NANP, and strip leading 1 off exten => _1XXXXXXXXXX,1,StripMSD,1 ; dial outbound on trunk group 1 exten => _XXXXXXXXXX,2,Dial,Zap/g1/${EXTEN} But when I upgraded on Friday to the latest CVSHEAD, this no longer works. If I send 13115552368 to this context, I get a message like pbx.c: Channel

Thanks to Greg Hill for pointing me to the 'sip debug on' cmd that helped me resolve the sip connection problem! The other issue I'm trying to resolve is configuring outgoing calls. I need to configure outgoing calls to use the FXO card in the PBX (zaptel device) via sip connected ip phones when a user dials 9. I need to support local and long distance dialing. Below is an excerpt of

Nevermind. Figured this out. I needed the following in extensions.conf to enable outbound dial. exten => _9.,1,Dial(Zap/2/${EXTEN:1},70,Tt) Thanks -----Original Message----- From: Info [mailto:info@psgsite.com] Sent: Tuesday, August 17, 2004 9:27 AM To: 'asterisk-users@lists.digium.com' Subject: RE: dialing out Thanks to Greg Hill for pointing me to the 'sip debug on'

Hi, I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card. When in Minicom, the only way I can dialout is if i issue ATS18=1 First. Otherwise I get a BUSY message. So thats fine. But when I dialout from asterisk, I get an immediate hangup, so my guess is that asterisk is not issuing ATS18=1 to the ttyI device. Here are my configs, any input would be greatly appriciated.

Hello, I am new to Asterisk, so RTFM answers welcome too (just include the FM's link :). I'd like to build a simple dialout server based on Asterisk. I installed 0.4.0 from package (a Debian SID machine, "server"). The client is gnophone (a Debian SID machine too, "client"). My modem is a GVC 56k voice modem connected to the server's serial port. I modified

I installed Asterisk and a digium wildcard (X100P). Using the extensions.conf with a few changes and a sip.conf file that includes two extensions, I can place calls between the SIP phones. I also can call in to the SIP phones from the PSTN using the X100P. On incoming calls I can hear the default demo announcement and call the digium IAX line. The main problem i'm having is calling out to the

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

Hello, I can not seem to be able to get StripMSD and Prefix to work for me in extensions.conf. This is an example of what I have: exten => _050.,1,StripMSD,1 exten => _50.,Prefix,01051 exten => _001051.,1,Dial(${TRUNK2}/${EXTEN}) exten => _001051.,2,Busy exten => _001051.,102,Busy What I want to achieve is to call 001051501657887 on TRUNK2 when dialing 0501657887. dialing

Hi, Having Asterisk-0.4.0 with 2FXO port and Samsung SKP 816H PBX in 2 offices. I am able to make call between two offices. But the problem is that call dosen't hangup. Office A [Asterisk+2FXO+SamsungPBX] <------------- I A X ------------> Office B [Asterisk+2FXO+SamsungPBX] Configuration files are given here.............. ------------------------ zapata.conf

On 2003-04-26 at 00:42, Jim Gottlieb (that's me) wrote: > [ccmenu] > exten=s,1,Ringing > exten=s,2,Wait,2 > exten=s,3,BackGround(5045) > exten=s,4,Goto,outtrunk|17005554223|1 ; if they just wait > exten=_X,1,Goto,outtrunk|17005554223|1 ; if they press 0-9 > exten=_*,1,Goto,outtrunk|17005554223|1 ; if they press * > exten=_#,1,Goto,outtrunk|17005554223|1 ; if they

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

I've been having trouble matching variable extensions on a zap channel (an E1 line). Doing it the extensions.conf way: [pri1] ; Match 8078078- calls include => m807nat include => m807mob include => m807oth [m807nat] exten => _80780782XXXXXXXXX,1,StripMSD(7) exten => _2XXXXXXXXX,1,SetVar,clidest=${EXTEN} exten => _2XXXXXXXXX,2,Goto(cli,s,1) [m807mob] exten =>

I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank