similar to: Error While Calling AGI

Hello All, Is there any way to pass additional parameters while calling AGI from *.call file? Channel: Local/1000 at from-internal MaxRetries: 0 RetryTime: 15 WaitTime: 15 Application: AGI Data: recordvoice.php Something like Data: recordvoice.php?id=3453&name=asterisk Cheers, Nitesh

I am using Realtime extensions as well, in case that would matter. Following problem arises from time to time, a call will successfully terminate: [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing [t at project_init:1] Hangup("SIP/peer-2-00002f7e", "") [May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init, t, 1) exited non-zero on

I'm doing some automated calling by putting .call files in the Outgoing folder of Asterisk. I'm concerned this might be a stupid question, but I'm pretty sure I've done my research well and I'm unable to come up with an answer on my own. I want to know: what happens to the .call files after the "MaxRetries" number has been reached? In my experience, they stay in the

Hello All, I got my TDM04B card installed and configured. Everything works fine I can receive calls and route to appropriate extensions. The only problem I am facing is Slowness. When I dial the PSTN number which is connected to Zap 1-1 after two ring it answers and then run the AGI script. What I did was assign it to a specific extension. So all inbound call on that PSTN number should

Hello All, I have my own AGI script running and I am trying to push the call to voice mail when Busy, Unavailable and Not Answered. Everything is working fine but the only problem is voice mail greetings for Busy and Unavailable is not played. By default only "Temp Greetings" voice mail greetings is played. I am passing the correct parameters for Busy => 'b', Unavailable

Hello All, Anyone using AGI scripts to monitor their systems? Something like if the system goes down, AGI script will be triggered and system admin will be notified saying "System XYZ has gone down"... Any suggestions... Cheers, Nitesh

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

Hello, you can have a look on Homer http://sipcapture.org/ regards On 27/09/2016 10:39, Gholamreza Sabery wrote: > Hello, > > For service monitoring you can use tools like sipsak in combination > with Zabix or Zenoss. Also using Zenoss or Zabix you can monitor the > health of your servers. This way you have both top-down and bottom-up > monitoring. For monitoring call

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To:

Hello all, The question isn't directly related to Asterisk, but I'm looking for recommendations for a monitoring tool to monitor the health of Asterisk instances running in Production. Ideally, the tool should be able to generate monitoring traffic (OPTIONS ping or INVITE), use the response/no response from Asterisk to store the health of an Asterisk instance running somewhere in the DB.

Hello All, Recently I added some Nokia N95 customers and it worked pretty good. Now the customers are complaining about the dialing rules... They are used to dialing +12486543210 and +4479XXXXXX for long distance calls. Is there anyway to create a "+" sign dial plan which will allow them to dial a number with "+" sign. Cheers, Nitesh

Hi Nitesh, Take a look at this http://www.microappliances.com/site/html/index.php?section=Products&page =clienthowto.php I've never implemented it though so I would appreciate some feedback on if it works. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Saturday,

I'm using chan_sip, I experimented with adding a 'Route' header in the originate command and used the Dial command like 'SIP/peer/exten', but problem is that Request-URI isn't populated correctly. I'm using Asterisk 13. Thanks, Nitesh On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <jcolp at digium.com> wrote: > Nitesh Bansal wrote: > >> Hello,

On 20.06.2017 18:44, Rik van Riel wrote: > On Mon, 2017-06-12 at 07:10 -0700, Dave Hansen wrote: > >> The hypervisor is going to throw away the contents of these pages, >> right? As soon as the spinlock is released, someone can allocate a >> page, and put good data in it. What keeps the hypervisor from >> throwing >> away good data? > > That looks like

On 20.06.2017 18:44, Rik van Riel wrote: > On Mon, 2017-06-12 at 07:10 -0700, Dave Hansen wrote: > >> The hypervisor is going to throw away the contents of these pages, >> right? As soon as the spinlock is released, someone can allocate a >> page, and put good data in it. What keeps the hypervisor from >> throwing >> away good data? > > That looks like

Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh

Hello, I'm using the following Dial command syntax: Dial*(SIP/peer/exten!sip:xyz at xyz.com <sip%3Axyz at xyz.com>*), the SIP URI after the '!' mark should be set as To-URI in outgoing INVITE from Asterisk. It works, but problem is that To-URI formatting is a bit messed up, It looks as follows: *sip:sip:xyz at xyz.com <sip%3Asip%3Axyz at xyz.com>*, it seems that Asterisk

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.