similar to: Includes include the includes?

I am grouping my extensions by building like so: 1XX is Building 1 2XX is Building 2 7XX is Office [Office] extensions has the following includes 7xx Include => Local Include => International Include => Building1 Include => Building2 [Building1] has 1xx Include => Office Include => Building2 Include => Local I don't want building1 to access international, but

Hi, we would like to build a gluster storage systems that combines our need for performance with our need for disaster recovery. I saw a couple of posts indicating that this is possible (http://gluster.org/pipermail/gluster-users/2010-February/003862.html) but am not 100% clear if that is possible Let's assume I have a total of 6 storage servers and bricks and want to spread them across 2

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer

I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP

I am newbie trying to setup about 12 Polycom Ip500's on an asterisk server. I am working on my extensions.conf and am trying to make it so that all my extensions can dial each other. My extensions are number 720, 721, 722, 723 ..etc in my from-sip context I began doing entries such as: exten => 720,1,Dial(SIP/720,20) exten => 720,2,Voicemail(u720) exten =>

You are right. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tarpo, Louie Sent: Thursday, August 11, 2005 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone You write out a

Could any of you provide me information on a good Polycom phone dealers to utilize. One who provides firmwares ..etc Thank you! Kenny ______________________________________________________ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/

I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. >From a SIP phone I can dial 700 and all other extensions. >From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I

We are testing our Asterisk server prior to deployment. The server has a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and one PRI for local calls. We are using sipp from two different stations routing a test number out the LD lines and another test number out the PRI line. We can not get more then 23 total active calls to connect to the test numbers, the test numbers

I'm having problems getting an Xorcom USB Bri 4 dialling out in Australia. I can receive calls into the system without an issue, but I can not for the life of me dial out of the system. Below are my configs, I'm hoping its something simple that I just can't see as I've been looking at it for to long. Can any one point me in the right direction. P.S. Yes it is meant to be in TE

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

svn trunk 31497 For the life of me, I can't get this :) I want to be able to catch the situation where the calling party hangs up *before* the call is connected to the called party. My dialplan is thus: macro DialExternal(exten) { Dial(Zap/G3/${exten},120,g,M(connected)); goto DialResult|r${HANGUPCAUSE}|1; Hangup(); }; But the goto dialresult is not executed: Executing

Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten => _811XXXX20,1,Goto(C-Internal,100,1) exten => _811XXXX21,1,Goto(C-Internal,200,1) [C-Phibee] exten => 100,1,Ringing exten => 100,2,Wait,1 exten => 100,3,Answer exten => 100,4,Dial(SIP/201&SIP/200,30) exten => 100,5,Hangup exten =>

Hi All, I have setup 2 asterisk servers using AAH 2.8. I have configured a IAX2 trunk between the 2 servers using the guide on dumbme. Trunk is not using register string and no authentication. In my dial plan I have 7XX numers on server B and 6xx numbers on Server A. Calls from my SNOM phones are ok between the extensions on the 2 servers. In server A I have a eicon 4 port BRI card connected.

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

Hi, I'm wanting to do a PCA on some data which is comprised of two different groups (to see how well the groups are discriminated). Is there a way to change the colour of the datapoints in a biplot so that I can easily see which group is which (eg objects 1-100, red, 101-200, black). Might be simple, but I'm new to R and can't seem to find how to do this. Thanks. Paul -- View this

Can someone tell me how to do this...Given the following line: exten => *97,3,VoicemailMain(${CALLERIDNUM}@default) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home =

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not