Displaying 20 results from an estimated 1100 matches similar to: "Help with dialparties.agi"
2009 Sep 17
1
Freepbx database
Hellos
I am using freepbx and asterisk.
I am writing an AGI script to edit the values in findmefollow table. The
script will enable users to delete and add follow me numbers from their
phones. I want it to enable users enable/disable follow me.
I can't seem to find a value in the database that deals with
enabling/disabling followme. Please help
--
Best Regards,
James Mutuku Ndeti
Agile
2009 Aug 24
1
Follow me IVR sounds
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 organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php
2009 Sep 22
1
setting up a IP based voip carrier account
Hellos,
My voip carrier has assigned me a IP based account...where they only give me
the IP to call through. I have setup the dial plan
exten => _7XXX.,1,Answer()
exten => _7XXX.,2,vmauthenticate(${CALLERID(number)})
exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y)
exten => _7XXX.,4,Hungup()
Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error
SIP/Y.Y.Y.Y-35dc
2009 Sep 08
1
Asterisk remote calls with low bandwith and high latency
Hello,
I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2
remote soft phones. The latency btw both sites is btw 500ms-700ms. I know
this is a shot in the dark...but are there ways of improving the voice
quality for the remote calls(btw site 1 and site 2), Other than increasing
bandwidth?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
2009 Aug 20
0
asterisk followme feature code
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
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer
2009 Sep 08
0
asterisk and link spa942 provisioning
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 management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
can help you achieve
2009 Aug 13
0
asterisk conference error/bug?
Hellos,
I am having issues with my meetme conferencing. When I dial the conferencing
number, It hangs after a few seconds.I have read somewhere that I need to
enable ztdummy, which I have done but still no changes.
Here is my log
~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~=
-- Executing [1;36;40mMacro [0;37;40m(" [1;35;40mSIP/1215-fc5b
[0;37;40m",
2009 Aug 12
0
meetme conference hangs in silence after dialing
Hellos,
I am having issues with my meetme conferencing. When I dial the conferencing
number, It hangs after a few seconds.I have read somewhere that I need to
enable ztdummy, which I have done but still no changes.
Here is my log
~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~=
-- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/1215-fc5b[0;37;40m",
2009 Aug 28
2
Help with call scenario
I am running asterisk and I want to achieve the following scenario
My goal in the end is to achieve the scenario (example using extension A and
Extension B)
1. Extension A has a line apperance of 4(4 calls can ring on it).
2. Extension B calls extension A(which is busy on one of the lines).
3. Extension A sees the second light blinking and hears the beeps (currently
working).
4. Extension B is
2009 Sep 02
0
problem with agi script not getting variable
I am learning agi scripting using php. I m using phpagi 2.x on asterisk 1.2.
I hve written a simple script that reads out the callerid using flite. My
problem is that I seems the script is not getting the callerID.
Bellow is the script
_________________
#!/usr/bin/php -q
<?php
/**
* @package phpAGI_examples
* @version 2.0
*/
set_time_limit(30);
2009 Sep 07
0
Freepbx database followme disable/enable value
Hello,
I am writing an AGI script to achieve the following
- Users can Disable/Enable followme from their extension. They can also
change the followme details from their extensions.
I have looked at the follow me table for freepbx. I can't see the field for
the values enabling/disable followme. Is this value stored in the database?
--
Best Regards,
James Mutuku Ndeti
Agile Systems
2005 Feb 17
2
arrgghhh dialparties.agi
Hi I've been looking for 10 minutes and cant find dialparties.agi
Can anyone tell me what folder this is located in as I'm going crazy.
(if it makes a difference I use asterisk@home and am replacing the AMP
dialparties.agi file)
Super big TIA,
Dean
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2005 Sep 01
1
dialparties.agi is returning no extensions to dial
Hi,
I set up a ring group. I would like for people who select a certain voice
menu option to ring a list of extensions (I have just one extension in there at
the moment) and if it doesn't answer to go to an extension's voice mail. I am
using a version of asterisk from CVS, last updated a couple of weeks ago.
This line in extensions_addtional.conf sends the call to ringgroup 3 if
2011 Nov 30
1
Installing asterisk on a server vs appliance(e.g digium mypbx)
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.
2005 Jul 19
0
When Incoming Caller-ID is Blank Dialparties.agi is shoving incoming IP Address into it.
Running Asterisk Head 1.0.9. Below is a trace of a call delivered to my system which had no caller ID. For some reason, dialparties.agi shoves the incoming provider's IP address into the caller ID so you never have a call that is screened for PrivacyDirector. Is anyone else seeing this issue as well? Have I missed a patch?
This call shows on the display with a name of "Unknown"
2007 Mar 26
1
Asterisk incoming caller id problem
Hi, guys,
For my server, if i use my handphone to call in the PSTN line by TDM400p
card, the server could not receive the caller id correctly. anyone knows the
problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is
as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of
my FXS zap extension created.
dialparties.agi: Starting New
2006 Feb 28
1
FW: Re: Delay on Phone ringing
Skipped content of type multipart/alternative-------------- next part --------------
asterisk1*CLI> soft hangup Zap/1-1
Requested Hangup on channel 'Zap/1-1'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
== Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
--
2009 May 16
1
Queue Load, Asterisk Disconnected
I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for
a queue up to 15 agents through a PRI line, it was working fine for more than 1
year, suddenly, when there is a load on the queue, the asterisk service
disconnects and the calls are dropped. And the service starts again after few
seconds, and so on.
I am not using fax.
I checked PRI by zttool and there are no alarms.
The cdr logs
2006 Jun 09
2
No CID on ZAP
I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs
line coming into a Digium TDM01B. It appears to not be getting CID at all.
If I hook up a POTS phone to the line CID comes through fine. Inbound and
outbound calls work fine but there is just no CID on inbound for this
channel.The incoming route for the channel is Zaptel Channel 0. No DID or
CID settings applied. My IP
2009 Oct 08
4
Dialplan problem
Hi people,
I have the following dialplan, but it doesn't have the behavior that I think it should have.
[default]
exten => 2001,1,Answer
exten => 2001,n,Dial(local/3005)
exten => 2001,n,Hangup
exten => 3005,1,Set(__RINGTIMER=10)
exten => 3005,n,Macro(exten-vm,novm,3005)
exten => 3005,n,Hangup
When I execute the Originate (AMI) with the argument Channel=local/2001, It rings