[asterisk-users] Is there a default dial plan that is not in extention.conf?

Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what

Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?

-users@lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in extention.conf? On Fri, Jun 25, 2010 at 02:25:38PM +0300, Eyal Goltzman wrote: Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according

[asterisk-users] unwanted entries created in dialplan from users.conf, how can I get rid of it?

Hello, When I call dialplan reload I can see the following lines: == Parsing '/etc/asterisk/extensions.conf': == Found -- Registered extension context 'default' (0x8a72410) in local table 0x8a679d0; registrar: pbx_config -- Added extension '_1XX' priority 1 to default (0x8a72410) . .

[asterisk-users] Where can I find a minimal set of empty configuration files (SIP only)?

Hello, After installing and learning Asterisk I found myself with a need for a minimal set of empty configuration files with only the must have stuff in order to setup a SIP only machine, is there a place to find it? Thanks, Eyal --

[asterisk-users] Using AMI Originate to call 2 outside numbers and connect them

Hello, Can I use AMI Originate to call 2 outside numbers (SIP) and connect them? How? Thanks Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Using AMI Originate to call 2 outside numbers and connect them

To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using AMI Originate to call 2 outside numbers and connect them On Sat, Jul 03, 2010 at 01:33:25AM +0300, eyal goltzman wrote: Hello, Can I use AMI Originate to call 2 outside numbers (SIP) and connect them? How? Originate one

[asterisk-users] SIP Trunk configuration problem - fromdomain

Hello, I'm trying to register to my provider sip trunk, I got from him an host IP (a.b.c.d) to connect to and my provider recognize me based on the fixed IP (x.y.z.w) he gave me (no need for username and password) In the sip.conf I add: [mytrunk] type=friend insecure=no host=a.b.c.d

[asterisk-users] How to change the IP in the SIP contact header

Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE

Re: [asterisk-users] How to change the IP in the SIP contact header

: Re: [asterisk-users] How to change the IP in the SIP contact header Have you tried setting externip= In the [general] of your sip.conf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman Sent: Monday, July 05, 2010

[asterisk-users] How can get user inputs from called party after dial?

Hi, I want to dial a party, play him a message and wait for his input, i.e. DTMF digits and use them to control the rest of the dial plan. How do I do it? If I use Dial it will not return until the end of the call, isn't it? Thanks, Eyal --

Re: [asterisk-users] How can get user inputs from called party after dial?

at 2:16 PM, eyal goltzman egoltz...@gmail.com wrote: Hi, I want to dial a party, play him a message and wait for his input, i.e. DTMF digits and use them to control the rest of the dial plan. How do I do it? If I use Dial it will not return until the end of the call, isn't it? Thanks

Re: [asterisk-users] How can get user inputs from called party after dial?

. -Bruce On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman egoltz...@gmail.com wrote: Thanks, but I'm missing something here, the dial command is where? I need to do something like: Dial(1234) Read(1 digit) DoSomthing(based on digit from 1234) And as far as I understand the Dial start

Re: [asterisk-users] How can get user inputs from called party after dial?

want with the variable ${numb} If any part of above is unclear to you, you must consult your friend, google, for examples of Asterisk dialplan. -Bruce On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman egoltz...@gmail.com wrote: Thanks, but I'm missing something here, the dial command