Re: [asterisk-users] All circuits are busy now on outgoing trunk call

2012-03-26 Thread James Mutuku
Yes they did. On 3/26/12, SamyGo govoi...@gmail.com wrote: Good to know, hope our replies did some help :) On Thu, Mar 22, 2012 at 7:39 PM, James Mutuku listmut...@gmail.com wrote: Hi, Thanks for the support. Issue solved. Somehow the routes on the fxo gw were not working. On 3/21/12

Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN

2012-03-23 Thread James Mutuku
are on 192.168.1.*Problem is I don't know how I can keep the PBX on this subnet, and also connect it via eth to the other vpn modem and give it another IP which is on a 192.168.200.* subnet. Any pointers?Thanks! -- Best Regards, James Mutuku Ndeti Agile Systems Limited

Re: [asterisk-users] All circuits are busy now on outgoing trunk call

2012-03-22 Thread James Mutuku
Hi, Thanks for the support. Issue solved. Somehow the routes on the fxo gw were not working. On 3/21/12, James Mutuku listmut...@gmail.com wrote: Hi, I have configured a route on the fxo to send all incoming sip traffic to the fxo ports. I will try set the specific digits and see

Re: [asterisk-users] All circuits are busy now on outgoing trunk call

2012-03-21 Thread James Mutuku
those traces it'd be more clear what is the issue from the carrier end rejecting the calls...Maybe your credit expired !! Regards, Sammy. On Wed, Mar 21, 2012 at 9:36 AM, James Mutuku listmut...@gmail.comwrote: I have setup a trunk on freepbx and the outbound route. Everytime I dial via

Re: [asterisk-users] All circuits are busy now on outgoing trunk call

2012-03-21 Thread James Mutuku
@192.168.9.250 Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.13 Content-Length: 0 -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer

Re: [asterisk-users] All circuits are busy now on outgoing trunk call

2012-03-21 Thread James Mutuku
Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales

Re: [asterisk-users] All circuits are busy now on outgoing trunk call

2012-03-21 Thread James Mutuku
dialled number. See your FXO gateway configuration Web-UI for outbound patterns OR verify that the FXO has its outbound line configured and working properly. On Wed, Mar 21, 2012 at 5:20 PM, James Mutuku listmut...@gmail.com wrote: I am setting up asterisk-fxo gw. 404 Not Found (User

Re: [asterisk-users] Installing asterisk on a server vs appliance(e.g digium mypbx)

2011-12-01 Thread James Mutuku
to them about their phones has been so-so. Historically we've had awful experiences with other Chinese phone vendors and have stopped considering products from Chinese companies. We did not actually try Yeastar products. On Wed, Nov 30, 2011 at 3:39 PM, James Mutuku listmut...@gmail.com wrote

[asterisk-users] Installing asterisk on a server vs appliance(e.g digium mypbx)

2011-11-30 Thread James Mutuku
Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially. From experience, what would be pro and cons for either option? -- Best Regards, James Mutuku Ndeti Agile Systems

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-12 Thread James Mutuku
http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship

[asterisk-users] Sending call information to handset

2009-11-22 Thread James Mutuku
to the handset Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve

[asterisk-users] setting up a IP based voip carrier account

2009-09-22 Thread James Mutuku
Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error SIP/Y.Y.Y.Y-35dc is circuit-busy Are there any settings I am leaving out? Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented

[asterisk-users] Freepbx database

2009-09-17 Thread James Mutuku
that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how

Re: [asterisk-users] Freepbx database

2009-09-17 Thread James Mutuku
I have tried all that. I just can't trace the value. this maybe the wrong list. I just thought someone might know On Thu, Sep 17, 2009 at 5:39 PM, cov...@ccs.covici.com wrote: Well, why not disable it from the GUI and see what changes -- this is sort of the wrong list, but maybe someone knows

[asterisk-users] Help with dialparties.agi

2009-09-10 Thread James Mutuku
Hellos, I have asterisk 1.2 and freepbx 2.3. I have edited the agi script(dialparties.agi). Everytime I restart asterisk, the file gets overwritten. How do I make sure my changes are not overwritten? What generates dialparties.agi? Thanks -- Best Regards, James Mutuku Ndeti Agile Systems

Re: [asterisk-users] Help with dialparties.agi

2009-09-10 Thread James Mutuku
I got my answer dialparties.agi is not generated. It get's copied from the core module when you 'Apply Configuration Settings' which is what allows it to be updated with new core modules. It is also copied when you reload asterisk James ___ --

Re: [asterisk-users] CDR Reporting

2009-09-10 Thread James Mutuku
I got it running a few days ago. I am using php4. Itshould run on php5 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users

[asterisk-users] asterisk and link spa942 provisioning

2009-09-08 Thread James Mutuku
Hellos, I need to send personal directory from asterisk to the ersonal directory of the linksys spa 942. Is this possible? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship

[asterisk-users] Asterisk remote calls with low bandwith and high latency

2009-09-08 Thread James Mutuku
increasing bandwidth? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve

[asterisk-users] Freepbx database followme disable/enable value

2009-09-07 Thread James Mutuku
followme. Is this value stored in the database? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can

[asterisk-users] passing commands asterisk cli and getting output using PHP AGI

2009-09-03 Thread James Mutuku
the output below on the cli debug AGI Rx EXEC show hints AGI Tx 200 result=-2 AGI Rx VERBOSE EXEC show hints returned -2 1 AGI Tx 200 result=1 From My understanding -2 means failure to find application What am I doing wrong? -- Best Regards, James Mutuku Ndeti Agile Systems Limited

Re: [asterisk-users] passing commands asterisk cli and getting output using PHP AGI

2009-09-03 Thread James Mutuku
I have included that but my scripts goes silent at AGI Rx EXEC Flite Hello 1215, you have dialed 1220. AGI Tx 200 result=0 Below is my script #!/usr/bin/php -q ?php set_time_limit(30); require('phpagi.php'); error_reporting(E_ALL); $agi = new AGI(); $asm

[asterisk-users] followme Script

2009-09-02 Thread James Mutuku
Hello, I am looking for a follow me script, where users can toggle follow me from their extensions and add follow me numbers from their extensions. Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization

[asterisk-users] problem with agi script not getting variable

2009-09-02 Thread James Mutuku
agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx EXEC Flite Hello, . -- AGI Script Executing Application: (Flite) Options: (Hello, .) -- Playing '/tmp/flite_buf_VTgzTg' (language 'en') As you can see, the callerID is not palyed out. What could I be doing wrong? -- Best Regards, James

Re: [asterisk-users] Help with call scenario

2009-09-02 Thread James Mutuku
I am new to AGI. I have written my first php agi script that gets the extension dialed and says it back the caller using flite. I am stuck on how to pass the comand asterisk –rx “core show hints to asterisk and get the data back. This isn’t the recommended way, but it does work: Let’s say

[asterisk-users] jitterbuffer for chan_sip on asterisk 1.2

2009-09-01 Thread James Mutuku
Hello, From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer, it says that there For Asterisk 1.2 there was no jitterbuffer in the RTP-based channels (i.e. chan_sip). I am using 1.2 and Ind there is no reason to upgrade. Are there any developments on this? -- Best Regards, James

Re: [asterisk-users] jitterbuffer for chan_sip on asterisk 1.2

2009-09-01 Thread James Mutuku
The project I am working on is really big. Unless I upgrade during christmas(by then the project will be several months overdue). Just googled further and saw some patches. I will try them and see. ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] jitterbuffer for chan_sip on asterisk 1.2

2009-09-01 Thread James Mutuku
I did am not the one who started the project. the client has been running 1.2 for years and they needed additional features set up ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona

Re: [asterisk-users] jitterbuffer for chan_sip on asterisk 1.2

2009-09-01 Thread James Mutuku
It's long gone. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Help with call scenario

2009-08-28 Thread James Mutuku
. Extension A sees the second light blinking and hears the beeps (currently working). 4. Extension B is notified that extension A is on another call. Where do I start...I have looked in the following call variables--$dialstatus and $devstatus but I can't get them working -- Best Regards, James

Re: [asterisk-users] Call variables(dialstatus?)

2009-08-27 Thread James Mutuku
I am running asterisk 1.2.9.1. I looked into devstate(). I still can't figure how to use it. Below is my script. I am dialing 1215,then I get a prompt on the state of extension 1215. I have set extension 1215 to busy(by making several calls to other extensions). I still get the prompt 1215 has a

Re: [asterisk-users] Call variables(dialstatus?)

2009-08-27 Thread James Mutuku
...the above post lacked some details I am running asterisk 1.2.9.1. I looked into devstate(). I still can't figure how to use it. My dial plan gives 1215 has state 0 on all scenarios. Below is my script. I am dialing 1215,then I get a prompt on the state of extension 1215. I have set extension

Re: [asterisk-users] Follow me IVR sounds

2009-08-26 Thread James Mutuku
thanks danny for the reply. I am looking into using flite to read out the prompts. if i ma ask...are there other voices other than the mechanical robotic male voice available for flite.? I have searched over the internet and I can't seem to find any ___

[asterisk-users] Follow me IVR sounds

2009-08-24 Thread James Mutuku
Hellos, I am looking for the sounds used in this ivr example http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me. The one with 6900. Any assistance is welcome. -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your

[asterisk-users] asterisk followme feature code

2009-08-20 Thread James Mutuku
Hellos, I have using asterisk 1.2 and freepbx 2.3. I need users to disable and enable followme from there phones. I can't see any support for it. Is this possible/available.? I have googled and I can't get information on it -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994

[asterisk-users] Individual PIN Code per Extension

2009-08-19 Thread James Mutuku
Hellos, I have astersist 1.2 working with freepbx. I want to tie pin codes to extensions(users). How do I do this? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management

[asterisk-users] Call variables(dialstatus?)

2009-08-18 Thread James Mutuku
call. Which variable do I use on my dial plan -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our

Re: [asterisk-users] Call variables(dialstatus?)

2009-08-18 Thread James Mutuku
and fourth callet that they are on extension is on another call. Which variable do I use on my dial plan -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management

[asterisk-users] asterisk conference error/bug?

2009-08-13 Thread James Mutuku
) exited non-zero on 'SIP/1215-fc5b' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1215-fc5b' -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer

[asterisk-users] meetme conference hangs in silence after dialing

2009-08-12 Thread James Mutuku
extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1215-fc5b' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1215-fc5b' -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your

[asterisk-users] Second voice caller notification

2009-08-08 Thread James Mutuku
Hello, I have asterisk and linksys ip phones setup. If an ext. is busy on phone, the ext. user is notified by a beep. I want to configure asterisk to notify(voice) the caller that the extension is busy on a call. Is this possible? -- Best Regards, James Mutuku Ndeti Agile Systems Limited

[asterisk-users] Asterisk+a2billing for over 10,000 ext

2009-05-13 Thread James Mutuku
Hellos, I want to setup Asterisk+a2billing for over 10,000 extensions for voip resale. Has anyone done this before. What are the hardware requirements and challenges? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] Asterisk routine maintenance activities

2009-04-21 Thread James Mutuku
Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy

2009-03-08 Thread James Mutuku
that variable in the case of the dial status being treated as BUSY. To get a 'busy' will involve single line phones, or disabling call waiting on the phone receiving the call. regards, PaulH James Mutuku wrote: Hellos, I want to configure asterisk so that if exten A transfers a call to exten

[asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy

2009-03-03 Thread James Mutuku
Hellos, I want to configure asterisk so that if exten A transfers a call to exten B, and B is either busy or the call is not answered, the call returns back to A. Is this possible? Please help James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department

[asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy

2009-03-03 Thread James Mutuku
Hellos, I want to configure asterisk so that if exten A transfers a call to exten B, and B is either busy or the call is not answered, the call returns back to A. Is this possible? Please help James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department

[asterisk-users] Intergrating vicidial with trixbox

2008-10-29 Thread James Mutuku
Hello, I am searched the net for tutorials on how I can Integrate vicidial with trixbox. I can't find any. Anyone who knows where I can get one? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya

[asterisk-users] asterisk appliance A50 vs asterisk open source + fxo cards

2008-08-01 Thread James Mutuku
Hi list, I need advice on which solution to implement, asterisk appliance A50 or just install linux on a pc and get tdm cards. Any comments? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya

Re: [asterisk-users] asterisk appliance A50 vs asterisk open source + fxo cards

2008-08-01 Thread James Mutuku
allow you to have redundant servers. External gateways would also be easier to scale when you need more lines. [/quote] 2008/8/1 James Mutuku [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi list, I need advice on which solution to implement, asterisk appliance A50 or just

Re: [asterisk-users] Help With dial plan

2008-07-22 Thread James Mutuku
is dialing something other than 3000 and that extension is not registered on your asterisk. just a wild guess. On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi list, Have installed trixbox and I am working with a fxo gateway to get

[asterisk-users] Help with dial plan

2008-07-21 Thread James Mutuku
Hi list, Have installed trixbox and I am working with a fxo gateway to get fxo calls to trixbox. I am using sip to send the calls from the gateway to trixbox. I have an extension 3000 on trixbox on [from-sip-external] on extensions.conf ,I have put the dial plan below. exten =

[asterisk-users] Help With dial plan

2008-07-21 Thread James Mutuku
Hi list, Have installed trixbox and I am working with a fxo gateway to get fxo calls to trixbox. I am using sip to send the calls from the gateway to trixbox. I have an extension 3000 on trixbox on [from-sip-external] on extensions.conf ,I have put the dial plan below. exten =

[asterisk-users] question about fxo cards

2008-07-09 Thread James Mutuku
Hi, has anyone worked with nxtvox(www.nxtvox.com) fxo cards? What is their quality? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:

[asterisk-users] has anyone worked with nxtvox fxo cards

2008-07-08 Thread James Mutuku
Hi, has anyone worked with nxtvox fxo cards? What is their quality? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant

Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards

2008-06-18 Thread James Mutuku
: Some customers are locked into two year contracts. That was the answer I got when adding four POTS lines to a system with four BRIs... Thanks, Steve Totaro On Tue, Jun 17, 2008 at 1:39 PM, James Mutuku [EMAIL PROTECTED] wrote: Michael, I agree. Here we use e1s(which have even more

Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards

2008-06-17 Thread James Mutuku
into the location? Michael On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote: Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry

Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards

2008-06-16 Thread James Mutuku
Brian J. Murrell wrote: On Sun, 2008-06-15 at 11:03 -0400, Steve Totaro wrote: On Sun, Jun 15, 2008 at 10:53 AM, Brian J. Murrell [EMAIL PROTECTED] wrote: On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote: Please advice on channel bank Dude. There's the cool new

Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards

2008-06-16 Thread James Mutuku
has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest

[asterisk-users] Please Advice on Best High traffic fxo gateway/cards

2008-06-15 Thread James Mutuku
Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards

2008-06-15 Thread James Mutuku
Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk

[asterisk-users] Asterisk Unified communication features

2008-06-12 Thread James Mutuku
Hi, I need to know if the following features are available on asterisk and their quality -SMS -Call control, budgeting and monitoring -Video conferencing -support for 500 extensions -fax -audio and video conferencing and 1. Call accounting showing calls made 2.