Hi,
Could someone kindly explain how does least recent strategy work?
According to the config:
leastrecent: rings the interface that least recently received a call
That does not explain much in detail. What happen if agent been idle (pause
member) or in wrap for length of time.. how does
Hi,
I am in process developing Multi-Tenant system for Call Centers.
I am considering what are the best option for Agent to Login and and wait
for the calls from the Queue.
Option 1: AgentLogin (staying on the line with music on hold and bridging
the call when a customer enters the queue)
Hi,
What options do I have to setup Distributed Device State across to multiple
Asterisk Servers?
If an agent is on the phone on a queue on one of the Asterisk server, other
servers will need to about it and therefore, will be able to operate
adequately. For instance, an agent is a member of two
Hi,
I am planning to move Asterisk from physical server to a VM on a ESXi host.
VMware datastore / VM's will be stored on the shared storage on the NAS
(NSF). I might get Synology NAS.
Do you store call live recording on the NAS? There would be around 60
concurrent calls recording at the same
is snippet for making QueueAdd request from AMI.
>
> -
>
> Action: QueueAdd
> Queue: supportqueue
> Interface: sip/1122
> Penalty: 1
>
>
> Regards,
> Muhammad Faheem
>
>
> On Tue, Sep 15, 2015 a
Hello,
Let say all the SIP devices will be registered on the proxy like kamailio.
Agent is a member of Support and Billings Queues on the asterisk servers.
Support queue on "Server A" and Billings Queue on "Server B" for example.
This will be done via RealTime Queue.
I want Agent to dial 1234
Hello,
Can someone recommend me where is best place to find Asterisk
Expert/Consultant for freelance work?
If you are interested to work as a freelancer, you can email me directly.
Thanks
--
_
-- Bandwidth and Colocation
, Richard Mudgett rmudg...@digium.com wrote:
On Fri, Aug 7, 2015 at 10:06 AM, Shahid H shah...@gmail.com wrote:
Hi,
If agents is already logged in via AgentLogin() and users dialled
extension 300 which will be placed in Queue(support-queue).
How to find out which agent is available I can
Hi,
If agents is already logged in via AgentLogin() and users dialled extension
300 which will be placed in Queue(support-queue).
How to find out which agent is available I can put their Agent id
in AgentRequest() ?
If this is not a good approach then how it should be done?
Agent should
Hi
Lately I've been having problem with the handsets on the polycom phones.
The agents are complaining that the humming/buzzing can be heard on random
days.
When the agents touch their computer the buzzing may increase louder or
reduced. Also when they grab the handset cable, the buzzing noise
Hello,
I am wondering has anyone used Live Recording (monitor or mixmonitor) on to
Storage Server via network 1 Gigabit connection?
Does it perform well, let say about 50 live recordings at the same time.
I am planning to make some system changes at work. I would like to put
Asterisk VM on a
I would like to develop a Call Center Dialer (outbound and inbound calls)
and it would use AMI method to communicate with Asterisk Server.
A daemon would need to run in the background, would you recommend coding in
PHP or Node.js? which would be much faster and stable.
Thanks
--
:
On Sat, Dec 28, 2013 at 11:32 AM, Shahid H shah...@gmail.com wrote:
Hi,
I would like to develop a Call Center Dialer (outbound and inbound calls)
and it would use AMI method to communicate with Asterisk Server.
A daemon would need to run in the background, would you recommend coding
in PHP
I wanted to create a daemon (background process) in PHP. A daemon will use
socket to connect with Asterisk AMI to send events and listen the actions.
A daemon will also listen the commands from agents via HTTP, for example:
A agent pressed a hang up button on a browser - it will send http
I wanted to create a daemon (background process) in PHP. A daemon will use
socket to connect with Asterisk AMI to send events and listen the actions.
A daemon will also listen the commands from agents via HTTP, for example:
A agent pressed a hang up button on a browser - it will send http
:* Saturday, August 04, 2012 7:34 PM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] Suggestion of Server Specifications for
Asterisk
On Sat, Aug 4, 2012 at 1:22 PM, Shahid H shah...@gmail.com wrote:
Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to
do 200
a 100 Mbit/s will be fine.
About UK provider, I can't be of any help... I know very good providers in
Germany and Canada, where I am laying my servers, but none in UK.
Leandro
2012/8/4 Shahid H shah...@gmail.com
What the minimum Server Specifications do I need to run
200 concurrent channels
...@gmail.com wrote:
It is not necessary to use an high performance drive. The bottleneck will
be the processor, not the disk. A single disk can handle ten times the load
of 200 ulaw channels.
Leandro
Il giorno 04/ago/2012 12:39, Shahid H shah...@gmail.com ha scritto:
Would a SSD drive be enough or do
Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to
do 200 calls recordings.
Once the call hangup/completed it will then move recording file to SATA HDD.
What do you think of this?
On Sat, Aug 4, 2012 at 5:51 PM, Benny Amorsen benny+use...@amorsen.dkwrote:
Leandro
What the minimum Server Specifications do I need to run
200 concurrent channels at the time with .WAV recording (MixMonitor)?
It will be connected via VOIP sip account.
Codec will be ulaw.
Which UK dedicated server provider do you recommend and how much bandwidth
do I need?
Thanks
--
I am having a problem with SendDTMF() - 50% of time it did not succeed.
I suspect it is not sending clear DTMF tones to the IVR.
For example:
SendDTMF(w3w2ww1w4)
Sometime digit 3 and 2 work, and failed to do digit 1.
Sometime digit 3 work and failed to do number 2.
Sometime
I am looking for a GSM Gateway or GSM PCI Card with minimum of 6 Sim Cards
slots.
Which one do you recommend and easier to setup?
As long it work on UK mobile network and make 6 calls simultaneously.
Thanks
--
_
-- Bandwidth
I am looking for a GSM Gateway or GSM PCI Card with minimum of 6 Sim Cards
slots.
Which one do you recommend and easier to setup?
As long it work on UK mobile network and make 6 calls simultaneously.
Thanks
--
_
-- Bandwidth
When I execute the ACTION commands set then the EVENT would response back.
How would I know which ACTION are they belong/reference to?
For example:
ACTION: Originate
Channel: SIP/test
Exten: 215
Timeout: 3
Context: test
Priority: 1
ActionID: 1333
Response: Success
ActionID:
the UniqueID.
What do you think of this solution?
Thanks
On Fri, May 11, 2012 at 2:31 PM, Matthew Jordan mjor...@digium.com wrote:
- Original Message -
From: Shahid H shah...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, May 11, 2012 6:12:25 AM
Subject: [asterisk-users
When I execute the Action commands set then the Event would response back.
How would I know which Action are they belong/reference to?
For example:
ACTION: Originate
Channel: SIP/test
Exten: 215
Timeout: 3
Context: test
Priority: 1
ActionID: 1333
Response: Success
ActionID:
I understand why do I get call twice to my mobile when I execute the
following AMI command sets:
ACTION: Originate
Channel: Local/800@test
Timeout: 6
Priority: 1
and my dialplan look like this:
[test]
exten = 800,1,DIAL(SIP/447xx@voip);
exten = 800,n,Hangup()
How to prevent getting
I am learning how to use AMI and I am having 1 problem.. When I make a call
to my mobile phone and when I answer it - it get disconnected/hangup right
away.
Why is that? What is the solution to stop that?
For example:
ACTION: Originate
Channel: SIP/447XXX@vpsprovider
Exten: 210
Priority: 1
timeout you are providing. Or it could be the
hangup() command in the 210 context.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Shahid H
*Sent:* Tuesday, May 08, 2012 3:11 PM
*To:* asterisk-users@lists.digium.com
, Danny Nicholas da...@debsinc.com wrote:
Since you are Originating the call, the hangup command isn’t needed.
Remove and reload.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Shahid H
*Sent:* Tuesday, May 08, 2012 3
When SendDTMF() finish the process then I want to hang up the call after 20
seconds.. What is the solution to do this?
I know there is S(x) option for Dial() application but it still count
during SendDTMF() process.
Thanks
--
_
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
I use software phone to test it... when I dialed 501, I cant hear anything
for about 10
:
Am 06.05.2012 13:46, schrieb Shahid H:
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
Log the actual DTMF to your console, set
most codecs.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
Sent: Sunday, May 06, 2012 9:16 AM
To: Markus
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
State: Early Media while it sending DTMF even
though I cant hear DTMF sound.. after 10 seconds State changed to Up (I
can hear talking to myself).
On Sun, May 6, 2012 at 4:18 PM, Shahid H shah...@gmail.com wrote:
When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF
sound.. completely
as
you like without validation?
I know voip.ms does it and sipgate don't allow it.
Thanks!
On Sun, May 6, 2012 at 5:08 PM, Shahid H shah...@gmail.com wrote:
Here is another debug log:
== Using SIP RTP CoS mark 5
-- Executing [123@test2:1] Dial(SIP/test2-0008,
SIP/+44776
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
I use software phone to test it... when I dialed 501, I cant hear anything
for about 10
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