Hi group,
We have realtime queue architecture on asterisk 1.8.7.0
I noticed that when we change strategy from any other to 'linear' it
requires Asterisk restart take the change in effect.
I have one realtime queue '1' with strategy set to 'ringall' and I change
its strategy to 'linear'. Now when
Hi,
Has anybody any idea about detecting Special Information Tone(SIT)
when making utbound calls?
http://en.wikipedia.org/wiki/Special_information_tone
I googled for detecting SIT in Asterisk but couldn't find useful results.
Thanks,
--Sam
--
that
this behavior is inconsistent, and thus, a bug, but I could see it going
either way.
If you open a ticket on the issue, respond here with the issue id, I'd like
to track it.
Thanks,
--Warren Selby, dCAP
On Oct 11, 2011, at 11:39 PM, Asterisk Man theasterisk...@gmail.com wrote:
Thanks Warren
...@selbytech.com wrote:
On Tue, Oct 11, 2011 at 12:58 AM, Asterisk Man
theasterisk...@gmail.comwrote:
snip
Event: QueueMember
Queue: 1
Name: 3
Location: SIP/
Membership: dynamic
Penalty: 2
CallsTaken: 0
LastCall: 0
Status: 5
Paused: 0
I would first troubleshoot why
Friends,
I was just playing with couple of manager actions for Queue statistics on
Asterisk 1.8.7.0 and found some inconsistency in information(I may be wrong
somewhere interpreting the information!).
Let me paste the outputs of my test for your reference.
=
Asterisk 1.8.3.2
I have been getting this warning constantly on CLI in a call scenario where
I use local channels to connect SIP with PSTN.
I use callfile and local channel to first call a PSTN number and if
answered, use local channel to call SIP phone with music on hold enabled in
Dial string.
Thanks for the response.
I have disallow=all and allow=alaw in sip.conf for my SIP user.
Any other idea?
--AM
On Tue, Jun 28, 2011 at 4:23 PM, Fellipe Paes fellipe...@hotmail.comwrote:
Hello!
In your sip.conf use alaw as your first codec option and see what happens.
Best regards,
Fellipe
Hi,
I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now
when a caller is placed into Queue and gets connected with Member, I want to
record the call. It does record the call when I use MixMonitor() before
placing the caller into Queue, but not when MixMonitor() is used in
:
On Thu, 2011-05-05 at 18:16 +0530, Asterisk Man wrote:
Hi,
I have a simple Queue(named 1) and one Member(SIP/1119) logged into
it. Now when a caller is placed into Queue and gets connected with
Member, I want to record the call. It does record the call when I use
MixMonitor() before placing
a realtime Queue in which members are added/removed dynamically.
Any help or pointer will be appreciated.
Thanks,
--AM
On Fri, May 6, 2011 at 9:52 AM, Asterisk Man theasterisk...@gmail.comwrote:
Thank you very much for your response and suggestion.
I raised the question because in my project I don't
Hi,
wrandom strategy for Queue says...rings random interface, but uses the
member's penalty as a weight when calculating their metric. So a member
with penalty 0 will have a metric somewhere between 0 and 1000, and a member
with penalty 1 will have a metric between 0 and 2000, and a member with
Thanks Jaron,
I understood the point from your explanation.
What should I do if I always want to ring a particular Queue member first
whenever he is available?
Yes, I can dial that member first before sending the call to Queue and
achieve the result but just wanted to know views from others.
Hi,
Can we know the estimated wait time for a caller before sending him in a
Queue?
Asterisk 1.8
Thanks,
--AM
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Hi Group,
Is there any information available for Asterisk 1.8 dimensioning? I googled
but couldn't find helpful data for 1.8.
I am trying to figure out hardware configuration for following features
implemented in Asterisk 1.8?
(1)100 SIP clients.
(2)ACD (Around 15 realtime queues)
(3)Call
Hi Group,
In Queue application, we have AGI,macro and gosub parameters that allow us
to perform some operations when Queue member gets connected with caller. But
it seems that right now there is no such mechanism (except CEL,AMI) for
situation where we want some operations to be performed when
Probably this will help you...
http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288901
Check the section 'Controlling when to join and leave a queue'.
--AM
On Thu, Mar 17, 2011 at 9:15 PM, Dan Journo
d...@keshercommunications.comwrote:
Hi,
I'm trying to work out an issue with
Hi Group,
I have following case scenario.
Through call file, Asterisk makes a call to SIP extension. When Extension
answers the call, Asterisk reads customer numbers (set in callfile) and
calls them one by one untill one of the customers answeres the call. Here
customer and SIP extension gets
...@lists.digium.com] *On Behalf Of *Asterisk Man
*Sent:* Thursday, March 17, 2011 8:13 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Answering machine detection for a second leg
callgenerated by a call file.
Hi Group,
I have following case
(MACRO_RESULT=CONTINUE)
So if an human is detected the legs will be bridged, if not the called
party will be hangup and the next number will be called.
The problem is, like previously said, the accuracy of the detection...
Best regards,
Federico
2011/3/17 Asterisk Man theasterisk...@gmail.com
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a
1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Asterisk Man
*Sent:* Wednesday, February 16, 2011 3:06 PM
*To:* asterisk-users
Hi Group,
Does Queue application take member penalty into account when strategy is
other than wrandom?
If yes, What difference does it make in case of linear and rrmemory
strategies?
Thanking you,
AsteriskMan
--
_
-- Bandwidth
Do you forward the call from SIP phone or Asterisk dialplan.
If it is from SIP Phone, above solution will not work. Infact any
solution will not work except your softphone supports call forwarding
based on some filter parameters.
--AsteriskMan
On 1/5/11, Danny Nicholas da...@debsinc.com wrote:
... atleast for the moment),
we should be little
bit more responsive.
Regards,
On Mon, Dec 27, 2010 at 4:35 PM, Asterisk Man theasterisk...@gmail.comwrote:
I need clarification on couple of issues of Realtime Queue.
It seems that when Agents(Memebers) are added using AddQueueMember,
Asterisk puts
I need clarification on couple of issues of Realtime Queue.
It seems that when Agents(Memebers) are added using AddQueueMember, Asterisk
puts this Queue-Member relationship information into AstDB, So that on
asterisk restart this can be preserved.
My question is, why does asterisk not store
A ton of thanks for useful information.
Quite informative to keep in mind for somebody like me who is still learner!
On Sun, Dec 26, 2010 at 5:21 PM, Sebastian s...@open-t.co.uk wrote:
Hi,
On 12/24/2010 12:37 PM, Asterisk Man wrote:
Friends,
Do we need to change any Asterisk
Friends,
Do we need to change any Asterisk configuration files (Or any file related
to Asterisk for that matter) when we put Asterisk box from one network to
another?
It is assumed that DB is on the same box.
Asterisk box has got Asterisk running in it with no issues.
Probably, it should not
Friends,
Do we need to change any Asterisk configuration files (Or any file related
to Asterisk for that matter) when we put Asterisk box from one network to
another?
It is assumed that DB is on the same box.
Asterisk box has got Asterisk running in it with no issues.
Probably, it should not
Christian,
Thanks for your response.
In my case, I was asked to do it through SIP phone 3 way call functionality
and not the Asterisk Meetme application.
I wanted to know if any one had done something similar in past or not.
I am short of PRI in my test environment and hence I can't test it
Will someone help/direct me find a way to implement this?
Or you can suggest some other method.
On Fri, Dec 17, 2010 at 12:44 PM, Asterisk Man theasterisk...@gmail.comwrote:
Hi friends,
I want to implement following scenario using Asterisk. Please suggest me
whether it is possible
the output.
'QueueSummary' didn't show any member logged into 'retailBanking', where as
'Queuestatus' did show members with 'QueueMember' event.
Is this bug or intended behavior?
Should I submit a bug report?
On Fri, Dec 17, 2010 at 4:59 PM, Asterisk Man theasterisk...@gmail.comwrote:
Asterisk
Hi friends,
I want to implement following scenario using Asterisk. Please suggest me
whether it is possible or
not.
This is bit off Asterisk and more on SIP side.
An Asterisk box with one Station(SIP channel) and PRI.
Agent dials a PSTN number of customer from station through Asterisk PRI.
Guys,
Why is such contradiction between 2 AMI actions QueueSummary and
Queuestatus?
Look at LoggedIn of QueueSummary and Event: QueueMember.
Also LongestHoldTime of QueueSummary does not give correct value.
-
Action: QueueSummary
Queue: retailBanking
Asterisk Version: 1.8.0
Members are added through AddQueueMember in realtime Queues
On Fri, Dec 17, 2010 at 4:52 PM, Asterisk Man theasterisk...@gmail.comwrote:
Guys,
Why is such contradiction between 2 AMI actions QueueSummary and
Queuestatus?
Look at LoggedIn of QueueSummary and Event
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