similar to: Bulk forwarding to another Asterisk

Dear all I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan I have setup asterisk with avaya system i have 5 avaya system on 5 location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when

Hia .... i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance

Is then new idmap_ad module capable of getting uid/gid info from a Windows 2003 AD pre-R2 with RFC2307 Unix Identity Mapping Extensions applied? Also, is the correct syntax for specifying the schema_mode as follows: idmap config dom.example.com:schema_mode = rfc2307 (I am not confident that I am reading the idmap_ad manpage and the new idmap document correctly.) Thanks for the help, Murthy

Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what.

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi All Has anyone used Asterisk for a Call Center operation? What I mean is: given a list of phone numbers, can Asterisk dial each number, play a message and accept some DTMF? I ask because I am an employee of a non-profit company based in San Diego, CA. I already evaluated Voicent and Voxeo. The former has expensive licensing terms and the latter is not best suited for a call center. I would

Hi All, I have the following setup. Samba Server: Scientific Linux 6.2 with the following samba config. I am able to ping the samba server from both a Mac and Windows XP machines. I can even connect to the samba share. However when I try to access files in the directory I get the error Access Denied. What am I doing wrong? Any help is appreciated greatly. From Mac I can Connect as Samba User,

I''m following along with the depot example in ''Agile Web Development'' and ran into a problem with Scaffold. I changed the MySQL table structure to add a few columns directly and also tried the migrate script to change the database after I got the initial scaffolding to work. However, I can''t seem to get the new columns to show up and am stumped as to how I can

Hi Guys, I am not so familiar with asterisk and hope to get help here. I am having now some stupid errors. My goal for the first, is to create a simple pbx with different context. As long as I use only the contex 'default' everything seems to work perfect. Now I tried to add another context i.e 'internal' and the asterisk is complaining for not finding the required extension in

I''ve committed documentation and a test suite for win32-mutex. Once again, I''m looking for a good sample program that we can use for the test.rb file that really demonstrates what you can accomplish with it. I scoured the web but didn''t find anything good. On another note, I came across another implementation called "fmutex" that I thought might be worth

There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks

Samba version: Samba 3.0 rebuild from SRPM on RedHat 8.0 w/XFS. I ran into this problem (which sounds a lot like Bugzilla Bug 51) when copying files from a CD-ROM to a Samba share on a W2K client. Double-clicking on the file name in an Explorer window gives the file not found error, but File->Open from Excel, and double-clicking the the filename in the file-open dialog-box works fine. The

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

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

Hello, Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? Thanks for any ideas. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Hi everyone, I have a fact which looks like that: drbdconnectionstate0. You see the "0" at the end... Now I''d like to access this fact in a "dynamic" way, the "0" should be variable. I''m thinking of something like that: if ${drbdconnectionstate0}${drbddevice} == ''Unconfigured''. But as you may guess, this does not work... Has

I am trying to set up a Grandstream GXV-3000 Video phone to Asterisk ver 1.2.21.1. The problem I'm having is that it can call other SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833

The following AMI command works well for all of the information I want: action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-00000028 Variable: CHANNEL(pjsip,XXXX) Where XXXX can be one of the many available items. However, when I create a call from A to B, all of the items return properly except: local_addr and remote_addr. More specifically, they return correctly for the A leg (that