Re: [asterisk-users] Problem with AGI Script

Hello, I had a similar problem with a PHP AGI script. I'm not sure if it's a bug or what, but it seems the new way of setting variables is an application, no way could I get it to work. In the end I set a user defined variable in the AGI like this: write("SET VARIABLE myvariable"); Then in t

[asterisk-users] Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?

over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use) --> Use IAX Can any one help me with some dial plan logic for this; I'm confused as to the best way around this? Thanks in advance Chris -- C

[asterisk-users] Meetme define context

Hi All, I'm still having trouble trying to figure out if it is possible to define (in the dial plan) a context for meetme? I'm using 1.4.4 with dialplan logic of: exten => 123,1,Meetme(,Msa,) This defaults to conferences defined within the rooms context of meetme.conf Is it po

[asterisk-users] Meetme context.

Hi All, Is it possible to specify the context of a meetme conference under 1.4.x? By default all meeting rooms are generated under the context rooms, I would like to use other contexts depending on what extension number is used to call the meetme application. If it is possible can some

[asterisk-users] RE: Zaptel linux26

Hi The various bits of instruction out there on compiling Zaptel on 2.6 seem to be a bit misleading. With the latest versions there is no need to run make linux26 Simply run Configure Make Make install Optionally I believe you can run make menuselect first to choose packages? Hope this help

[asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS

from CD my self). Again, many thanks Chris -- Chris Blunt -Original Message- Date: Tue, 10 Apr 2007 19:56:43 +0300 From: Tzafrir Cohen <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS To: Asterisk Users Mailing List - Non-Commercial Discussi

[asterisk-users] Re: Zaptel 1.4.1 Install Modules CentOS

I can't thank you enough for your continued help. Chris -- Chris Blunt -Original Message- yum install kernel-smp-devel > > I did check the "/lib/modules/2.6.9-42.0.3.ELsmp" directory but there is no > build link, could this be the problem? Yes. No suggested loca

[asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS

t: 2.6.9-42.0.3.ELsmp) and did "yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686" If I now do ls -l /lib/modules/`uname -r` I do get " build -> /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686" I have then tried recompiling zaptel. But same trouble I'm afraid! I can'

[asterisk-users] RE: asterisk-users Digest, Vol 33, Issue 15

-42.0.3.EL kernel-utils-2.4-13.1.83 I did check the "/lib/modules/2.6.9-42.0.3.ELsmp" directory but there is no build link, could this be the problem? Again thanks for your help, I am only a Linux beginner, and even more of a noob with CentOS. Best regards Chris -- Chris Blunt ---

Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS

harset=us-ascii On Tue, Apr 03, 2007 at 11:57:57AM +0100, Chris Blunt wrote: > Hi All, > > > > I have a CentOS server that I am trying to configure Asterisk on 1.4 on. > > > > Everything seems to go ok, with regards to compiling Zaptel, Libpri, >

[asterisk-users] Zaptel 1.4.1 Install Modules CentOS

y don't know what is going wrong with the modules? I'm a bit out of my depth with CentOS, as this isn't my server (I'm a Slackware guy) Any pointers seriously appreciated. Thanks Chris -- Chris Blunt __

[asterisk-users] Sipura SPA2000 Transfer Call

-- Chris Blunt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Problems with CentOS ztdummy kernel 2.6

ols/genmodconf linux26 "" "tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc wct4xxp wcfxs wctdm8xxp wct2xxp" Building /etc/modprobe.conf... Once it is installed I run: modprobe ztdummy with the following result. FATAL: Modul

[asterisk-users] Need quality toll free 800 number over IAX?

Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided by Easynew

[asterisk-users] Agent autologoff dynamic queue members - Brain aches please help

Hi list, Using Asterisk 1.2.10 I am getting seriously confused by Queues and Agents. So far I configured my queue and agents, had my agents login using agentcallback. Call enters queue agent gets a call, if agent doesn't answer after 20 seconds a flag is set in AstDB (thanks to: Leo A

[asterisk-users] Help with dial plan - two attempts at calling agent before logging agent off?

Hi List, I'm attempting to set up a queue and agents using agent call back. This is all working fine with the queue and the agents login etc However. In my dial plan I a set variable when a call is entered into the queue to identify the origin of the call, then when the agent is call

[asterisk-users] AGI PHP Issues (AGI script runs but phone hangs up too quickly)

Sorry to re-post this but I'm sure it's something simple that someone has found before. To summarise: Dial plan answers the phone AGI script executes AGI debug in console show phonetics ABC - However no audio at all on the phone and this step is less than 1 second. Dial plan Busy Pho

[asterisk-users] AGI PHP Issues (Not new to Asterisk but new to AGI)

Sorry to bother you all with what is probably a simple question. I am attempting my first go at a simple AGI application using PHP (Getting Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means a professional standard developer. My script seems to execute ok, and I ca

[asterisk-users] Is there a smarter way to ban expensive calls in dial plan?

Best regards   Chris   --   Chris Blunt Entropy IT Ltd   ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Load balenced (ADSL) network connections, is it possible?

each with a different IP address pointing to a different default gateway (router). But then some how load balanced into a virtual network connection?   Any ideas or solutions would be appreciated – just in case I have gone off at a wild tangent.   Thanks   --   Chris Blunt Entropy IT Ltd

[asterisk-users] Dial plan question

    Thanks for your time and advice.     --  Chris Blunt   ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Upgrading

overwriting any modified sound files etc?  Should I delete the current files or move / make a copy to a different location first?   I know this is a lot of questions but I am hoping for a best practice idea etc…   Regards   Chris   --   Chris Blunt Entropy IT Ltd

[Asterisk-Users] RE: Configure Voipjet.com content in Asterisk

Hi Chandramouli Setting up VoipJet is quite simple really, you have done all the hard bit to get you Asterisk config this far. Firstly may I point out if you are posting your configuration to this list you change your password information, as you have just given everyone access to your account a

[Asterisk-Users] Meetme from MySQL

Hi List,   Is it possible to store meetme config in a MySQL table?   If so, any pointers would be appreciated.   Thanks   Chris     --   Chris Blunt Entropy IT Ltd   ___ --Bandwidth and Colocation provided by Easynews.com

[Asterisk-Users] Looking for quality inbound DID - IAX providers, UK, USA, Australia

Hi All,   I am looking for a provider/s of inbound DID – IAX numbers, for UK, USA, and Australia.   Preferably free or low cost J   Can anyone make a good reference?   Many thanks   Chris   PS: I appreciate this is perhaps a little OT, please feel free to reply off list.

[Asterisk-Users] Transferring calls into MeetMe

achieve this, without having to make two calls, transfer them in, then connect my self.   Any help or insight really appreciated.   Best regards   Chris Blunt --     ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Three way calling with X-Lite / MeetMe

is done using a feature of the phone, and X-Lite doesn’t look like it supports this.  Can this be achieved with MeetMe or AppConference, if it can please tell me how J   Many thanks   Chris   --   Chris Blunt   ___ Asterisk-Users mailing

RE: [Asterisk-Users] Digium Cards connecting to BT

Hi, There are several people on the UK mailing list (I am one) that have purchased the TDM400P FXO and are having problems with disconnect. Basically the cards are great (sound quality etc) but give some issues with detecting a UK remote hang-up. Mainly an issue within IVR, MeetMe and VM. There a

[Asterisk-Users] TDM400P FXO - Any one got it working well in UK without Hangup problems

Hi Guys,   I recently got a TDM400P 4 FXO for use in the UK, this at the time seemed like a good idea as I had good results with an X100P clone.    Installation went great and call clarity is excellent no echo like I had on the clone card.   My problems start with detecting hanging u

RE: [Asterisk-Users] # Transfers.

Thanks to Bruce for adding this stuff on attended transfers to the WIKI pages. I've been trying to get my head round this for a couple of days. Unfortunately I'm still having a bit of trouble. I have the latest CVS-HEAD, just downloaded and compiled. Added the bit for attended transfer into the

RE: [Asterisk-Users] Problem with demo on asterisk

I'm by no means an asterisk Guru, just trying to get is together my self. How ever, no sound issues usually relate to blocked ports on your router / firewall. If your extension 1000 is an IAX connection, check your rtp.conf, and perhaps narrow the port range, allow port forwarding on this range (

[Asterisk-Users] Attended call transfer

appreciated.   I have been on an almost vertical learning curve with Asterisk and Linux for 6 months this is just about my last challenge (for now – haha).   Many thanks   Chris Blunt   --   SIP: [EMAIL PROTECTED]     ___ Asterisk-Users mailing

[Asterisk-Users] Grandstream BT100 -> Asterisk -> Voipjet ..... No ring ring when making a call

Hi All,   I’m sure this is something simple that I have missed somewhere.  When I make a call using BT100 over IAX2 with Voipjet terminating I don’t get a ringing sound whilst I’m waiting to be connected.  The destination party can answer the call (they do get ringing) and conversation c

RE: [Asterisk-Users] Echo - UK Impedance problem with X100P?

Hi Soren, Thanks for your reply on this. My card is a clone, with an Ambiant 3200 chip. The parameter you gave me has sorted out many of my problems. It is people such as your self who are incredibly helpful within the Asterisk community. As like many others, I am relatively new to Asterisk

RE: [Asterisk-Users] Echo - UK Impedance problem with X100P?

Thank you to all that have posted so far. I realize the X100p clones are designed as voice modems. But if they are designed for the UK market and are BABT / EU approved, should they not support UK impedance? If these clone cards were capable of multiple impedance settings, how do we change Aster

[Asterisk-Users] Echo - UK Impedance problem with X100P?

, but found nothing yet.   Any pointers appreciated.   Regards   Chris Blunt   --   SIP: [EMAIL PROTECTED]       ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] Anyone using Asterisk on Slackware 9?

Hi, I am trying to do a very minimal install of Slackware to run Asterisk on.   Can anyone give me a list of what packages I need to install as I don’t want X an all the associated bloat?     Thanks in advance…   Chris   --     ___

RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

/wiki-Asterisk+firewall+rules http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD HTH Ed From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt Sent: 15 August 2004 23:06 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound

RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

t   [default]   ;inbound dialing from FWD exten => ${FWDNUMBER},1,Goto(housemenu,s,1)  ; I have mine set to hit a menu, no reason you cann't forward to an extension instead   - Original Message - From: Chris Blunt To: [EMAIL PROTECTED]

[Asterisk-Users] Inbound Free World Dialup - extension not ringing?

60) exten => 232999,1,Dial(SIP/phone1,30,tr) exten => 232999,2,Hangup     I am behind a NATed fire wall, but I’m not sure that is related.   Any ideas or help (working simple confs) would be much appreciated.       Best regards   --   Chris Blunt   SIP: [EMAIL PROTECTED]