similar to: AGI execution after Dial

Hi, I have a problem dialing a SIP phone which is logged in as different Astesrik machine from the one I am working with. I want to call a phone in Another astersik machine in , if it answers, calling a SiP phone registered in my ASterisk: My dialplan is: [mariaSIP] exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/6021@192.168.0.51:5038,20) exten => _1.,3,HangUp() exten =>

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at lists.digium.com >

On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > <snip> > >> Here

On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:55:28 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > > > > > On

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

Hi, I'm seeing a very strange error when dealing with Diversions. If a call setup to a number comes to an Asterisk server, that server sends a request to a third proxy, that proxy sends the call back with a Diversion flag, Asterisk complains about the host not existing (and the host is the number). Here's the output from the Asterisk CLI with SIP debugging enabled: <--- SIP read from

Okay, so I have an Active Directory server running on Windows Server 2012 Standard I have configured Samba/Kerberos/Winbind on Ubuntu 13.04 to bind to the DC properly. I am able to login with my Active Directory users credentials. When I use the 'require_membership_of' option in pam.d/common-auth for winbind.so using the SID of the group I want to restrict access to, it works like a charm.

Group I have a Cisco 7970 Running the newest SIP image. I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC When I get a call the callerid number show something like 5555555@192.168.1.2 I thought I seen somewhere what that was but I'm unable to find the correct wording when searching Google to find that post again. Can anyone help me out here. How can I remove the asterisk

I've got this in extensions.conf: [macro-stdexten] exten => s,1,Dial(${ARG2},30,p) exten => 6015555555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+15045555555) Where the real numbers have been replaced with 5555555. What I'm trying to do is ring my cell phone in addition to the local extensions. Funny thing is the cell phone rings

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following: exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70) There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the

Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111.222 at mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is < sip:111.222 at mydomain.com> and From header has value "username" < sip:111.333 at mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends out the

Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question Aaron Yes it

Hi, I am searching for the most effective solution for the following scenario: Our users can call into our IVR menu and dial a specific extension and immediately hang up. This event should simply trigger Asterisk to make multiple simultaneous calls through a group of zap channels (5-10 calls). When the called parties answer, Asterisk should simply play a message and hangup. So I was thinking

Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olli Heiskanen Sent: 03 January 2015 08:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk removes a charachter from sip peer name Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names

Hello to everyone, I have a problem that is giving me a headache, and trying to do a project in Ruby on Rails and I need to connect with a Data Base that is en Postgres, the truth is that I didn’t think that it was so hard to connect a DB with postgres because I was working with Msyql and everything was easy. This are the thinks that I have install in my computer. 1. Debian GNU/Linux, kernel