similar to: Example config for SPA-1001

I have a TDM4xx card with two (3 and 4) interfaces for my land lines. I have a basic setup working with them and one VoIP provider. Questions: 1. How do I determine which Zap line the incoming call is on so I can handle it differently? One line is my home phone and the other is my work line. I would like different dialplans for each. 2. When I have my work line set (via Verizon) to call

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i

I'm trying to set up my extensions.conf file using some of the existing macros like stdexten, etc. while at the same time having two logically separate virtual PBX's (with no "default" context) and two trunks coming into separate contexts, i.e. one for residence and one for my at-home business. I noticed, however, that macro-stdexten depends on the "default" context:

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

with the below defined in logger.conf on 11.6 cert 6 I am not getting any log message other than notice and warning in any files when doing module reload logger - queue log is the only one that says it restarts *CLI> module reload logger == Parsing '/etc/asterisk/logger.conf': Found Asterisk Queue Logger restarted built fresh box with make samples - added 2 stations, dialing from

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly hear music on hold and then it stops until the timeout for no answer

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten =>

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

Dear All I want to integrate sugarcrm and asterisk , so when customer call the call center the agent or extension which answers the call , before pickup the phone and talk to customer , view his/her information if it is available. I do this as described below : 1-Setup login username for sugarcrm for each extension 2-Extension Users will login to the sugarcrm. 3-Develop php script to handle new

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

I've been banging my head on a brick wall for about an hour now trying to understand why the following doesn't work (which is even provided as an example in the distribution!). The goal is to create a voicemail-only extension not associated with a phone. I'd rather not have an extension dedicated to VoicemailMain(), so I would like the user to be able to hit '*' during

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf: