Re: [asterisk-users] Best way to update ever changing dialplans

[Lenz Emilitri]

Depends on how complex the generated dial-plan is.If it is in a
monitored production environment, you could have an ARI app and just
let it handle it all, especially if it needs to make complex decisions
that end in simple choices (like: detecting the caller, asking them
some questions but finally routing to queue A versus B).

2018-06-25 18:54 GMT+02:00 Dovid Bender :
> I am working on a system where I connect to an external API and based on
> what it gives me I generate the Asterisk dial plan accordingly. I am
> thinking about my different options and wanted feedback from others on how
> to best do it.
> 1) Generate conf files for Asterisk - This seems the easiest but then I will
> be doing a dial plan reload on all of my dial plan for handful of lines of
> code. The plus side is once reload is don the dial plan is in memory.
> 2) Using real time + mysql - Seems like an overkill to have mysql running
> taking resources for a few lines.
> 3) Using real time + sqlite3 - This seems like the best option but then we
> go to disk every time there is a call.
>
> Any other options that I am not thinking of?
>
>
>
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>
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[asterisk-users] A survey on Asterisk-based call-centres - Help needed

[Lenz Emilitri]

Hi all,
I am running a survey of Asterisk-based call-centres, to understand
what they are doing now and how they expect to grow in the future.
Results will be presented by yours truly in October at the Astricon in
Orlando, but you can also sign up to receive them when they will be
ready. See 
http://www.digium.com/blog/2018/06/27/asterisk-contact-center-survey-results-will-be-interesting/

So, if you run a call-center based on Asterisk, or you have customers
doing it, why not letting the community know what you are doing and
what you wish for? it only takes 5 minutes, but it’s a way for your
voice be heard.

You can find the survey at:
https://www.queuemetrics.com/callcenter-survey-2018.jsp?lid=B083

Best,
lenz


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[asterisk-users] Tracking Music-on-Hold on call queues

[Lenz Emilitri]

Hi all,
we have a little tool that tracks Music-on-Hold events for call queues
by listening to AMI events.

This is quite useful for reporting so, as the tool is free to use and
does not depend on our QueueMetrics Call Center suite, I thought I'd
announce it in here as well.

If anyone is interested, you can find a post here:
https://www.queuemetrics.com/blog/2017/03/22/TrackingMOH/?lid=A002

Comments welcome :)
l.






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Re: [asterisk-users] Pet project: one step Asterisk compile on Centos 7

[Lenz Emilitri]

2016-06-14 17:44 GMT+02:00 Tzafrir Cohen :
>
> 1. Asterisk basically has such a script inside.

It is - as you say - inside. This is outside and does the download for you.

> 2. Asterisk has an RPM package. An RPM package is exactly a reproducible
> build (listing dependecies, and such).

It's true. They are very interesting, especially if you are a
historian of software.
http://packages.asterisk.org/centos/

If you need something less, say, "vintage", you may need to compile it yourself.

> 3. You are reinventing RPM. Badly. Do you people really want to run:
>- As root
>- A huge blob nobody can inspect
>- that is executable, and hence has tons of places to add nice hooks
>  in?
>
> Learn how to use rpmbuild.

I personally happen to have shipped RPMs for about 10 years now. But
building a RPM might be overkill if you are deploying a test,
throwaway box, or just once for a Docker image. Of course I would not
use this as an RPM substitute, and if I were to use something like
this I'd fork it or at least read it (it is maybe 20 lines). YMMV.

And IIRC there is more places to ship "nice hooks" into a binary you
ship as an RPM than in a  shell script that does what you would
manually from the terminal!

l.




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[asterisk-users] Pet project: one step Asterisk compile on Centos 7

[Lenz Emilitri]

Hi all,
I thought I'd share I script I made (based on some of Leif's works)
that lets you download, compile and install Asterisk all in one go;
and then removed the dev tools used.

We use it quite a bit to provision systems using Ansible, but it is
easier than remembering everything every time even if you are using a
shell.

At the moment I have scripts for Centos 7 and Asterisk 13, but plan to
port  them to other versions of Asterisk as there is a need to do so.
Contribs welcome!

Project located at https://github.com/l3nz/CompileAsteriskPBX

Thanks
l.





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Re: [asterisk-users] [SOLVED] AMI issue with Filter

[Lenz Emilitri]

D'oh moment - filters do work, but you can only set one at a time. By
reading the docs, I mistakenly assumed that you could set many of them
at once.
Thanks
l.


2016-05-19 12:54 GMT+02:00 Lenz Emilitri <lenz.lo...@gmail.com>:
> Hello all,
> I am trying to use the Filter action in AMI to make AMI less chatty by
> blacklisting some events; and I must be doing something wrong, because
> if I send something like:
>
> Action: Filter
> ActionID: AID563116752-152218
> Operation: Add
> Filter: !Event: VarSet*
> Filter: !Event: ExtensionStatus*
> Filter: !Event: NewAccountCode*
> Filter: !Event: NewCallerid*
> Filter: !Event: Newexten*
> Filter: !Event: RTCPSent*
>
> I still get plenty of:
>
> Uniqueid: 1463577738.539
> Extension: 201
> Channel: SIP/200-00a1
> Context: from-internal
> Event: Newexten
> Application: Set
> Privilege: dialplan,all
> AppData: ADMINCODE=15
> Priority: 1
>
> I get them with or without the trailing *.
>
> I am testing this on Asterisk 11 and 13, so I must be doing something
> wrong - but what? :-)
> Thanks
> l.
>
>
>
>
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Re: [asterisk-users] Call File - CPU spikes

[Lenz Emilitri]

If you are on 13 it would likely be easier to use ARI directly?
l.


2016-05-11 22:52 GMT+02:00 Bryant Zimmerman :
> I am working on a project that we are seeing a 100% CPU spike when we move
> 50 calls files to the folder.
>
> We are running pjsip and asterisk 13..It holds the spike for several minutes
> Are there any tunable that may help with this?
>
>
> Thanks
> Bryant
>
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[asterisk-users] AMI issue with Filter

[Lenz Emilitri]

Hello all,
I am trying to use the Filter action in AMI to make AMI less chatty by
blacklisting some events; and I must be doing something wrong, because
if I send something like:

Action: Filter
ActionID: AID563116752-152218
Operation: Add
Filter: !Event: VarSet*
Filter: !Event: ExtensionStatus*
Filter: !Event: NewAccountCode*
Filter: !Event: NewCallerid*
Filter: !Event: Newexten*
Filter: !Event: RTCPSent*

I still get plenty of:

Uniqueid: 1463577738.539
Extension: 201
Channel: SIP/200-00a1
Context: from-internal
Event: Newexten
Application: Set
Privilege: dialplan,all
AppData: ADMINCODE=15
Priority: 1

I get them with or without the trailing *.

I am testing this on Asterisk 11 and 13, so I must be doing something
wrong - but what? :-)
Thanks
l.




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[asterisk-users] Sample Docker images for Asterisk available

[Lenz Emilitri]

Hello all,

I created a set of Docker images running Asterisk and exposing AMI /
ARI ports that i found to be quite useful for ARI / AMI development
and regression.

As they are based on Docker with whaleware, adding new configuration
files to roll your own dialplan / queues / voicemail etc is pretty
easy. And you can run quite a lot on the same box to simulate
clusters.

There is no SIP / RTP configured at the moment.

See 
https://github.com/l3nz/whaleware/blob/master/examples/asterisk-load-test/README.md

Maybe somebody else might find them useful. There is Asterisk 1.8, 11,
12 and 13.
Thanks
l.




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Re: [asterisk-users] Sample Docker images for Asterisk available

[Lenz Emilitri]

If you need to add files, either you create a new Docker image that
inherits from one of the images and add properties files to
/ww/files, or just copy the files into a docker instance and do a
docker exec to have Asterisk reload them. See e.g.
http://stackoverflow.com/questions/22907231/copying-files-from-host-to-docker-container

Make sure you have a look at what whaleware does, as it acts as a good
template and manages a number of things for you (eg configuration).

Just .02/chf
l.



2015-04-23 8:47 GMT+02:00 Guenther Boelter gboel...@gmail.com:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 On 04/23/2015 02:03 PM, Lenz Emilitri wrote:
 Hello all,

 I created a set of Docker images running Asterisk and exposing AMI
 / ARI ports that i found to be quite useful for ARI / AMI
 development and regression.

 As they are based on Docker with whaleware, adding new
 configuration files to roll your own dialplan / queues / voicemail
 etc is pretty easy. And you can run quite a lot on the same box to
 simulate clusters.

 There is no SIP / RTP configured at the moment.

 See
 https://github.com/l3nz/whaleware/blob/master/examples/asterisk-load-t
 est/README.md

  Maybe somebody else might find them useful. There is Asterisk 1.8,
 11, 12 and 13. Thanks

 Great, will try it out tonight ...

 Thanks


 - --
 DavaoSOFT, the home of ERPel
 ERPel, das deutsche Warenwirtschaftssystem fuer LINUX
 http://www.davaosoft.com
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[asterisk-users] Tutorial: compiling and installing Asterisk 13

[Lenz Emilitri]

Hi all,
I just prepared a little tutorial on installing Asterisk 13 on CentOS
6.5 64-bit.

See http://astrecipes.net/index.php?n=668

Hope you like. :)
l.




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Re: [asterisk-users] pull a call from a queue

[Lenz Emilitri]

What you usually do is to transfer the call to a second VIP queue.
This can be done in  the free version of our QM or I'm sure there are
other products as well :)

2014-06-13 20:15 GMT+02:00 Adam Moffett adamli...@plexicomm.net:
 We have a queue monitoring application running so we can see the caller ID
 of callers in a queue.  If we see a VIP in the queue, is there any method to
 force that call to be first in line?  If there's a softphone, or queue
 managing application already written that does this, I'd love to know.


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Re: [asterisk-users] Open Source Asterisk Polling Solution

[Lenz Emilitri]

Our Wombat is not open source but is free to use for small systems and
will be very trivial to set up for such a task. And if you ever need
to grow up, you're covered. :)
l.


2014-04-21 19:45 GMT+02:00 Nick Cameo sym...@gmail.com:
 Hello Everyone,

 We are looking for a simple open source auto dialer with polling
 capabilities. What we would like is a program that we can upload
 leads to, and have asterisk:

 i) Dial numbers
 ii) Play pre-recorded
 iii) If user presses one, forward the call to an agent

 There are so many solutions out there it's hard to make a decision on what
 works, what has just a limited free version etc Something that can
 support
 10 channels, and is stable would be greatly appreciated.

 If this can be simply implemented using asterisk and call folder, even
 better

 PS Our preferred version of * is 1.8.x

 Kind Regards,

 Nick from Toronto.

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Re: [asterisk-users] Numbers hackers call

[Lenz Emilitri]

http://en.wikipedia.org/wiki/Telephone_numbers_in_Israel

Looks like it a mobile in Palestine -  sure someone from Israel can
tell us more

2014-03-26 16:05 GMT+01:00 Michelle Dupuis mdup...@ocg.ca:
 I see a lot of attempts by hackers to call 00972595301123 or 011972595115207
 or variations but that same 972595 is often present.


 Can someone break down that dial string with an explanation?  The 011 look
 like an overseas call (from Americas), while the 972595XX is unclear...


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[asterisk-users] Couple of new tutorials on asterisk 12 and ARI

[Lenz Emilitri]

Hi all,
I put together a couple of new tutorials on compiling Asterisk 12 with
PJSIP  on CentOS 6.5  and test-driving ARI on the same box.

You can find them at:

http://astrecipes.net/index.php?q=AstRecipes/Compiling%20Asterisk%2012%20on%20CentOS%206.5

and

http://astrecipes.net/index.php?q=AstRecipes/Getting%20started%20with%20ARI


Comments welcome and happy holidays! :)
l.









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Re: [asterisk-users] Call Queue advise

[Lenz Emilitri]

Most likely the feature can be obtained with the wrap-up time or
pausing the agent. I agree on removing ring-all if possible (Though a
number of clients want it in smaller set-ups, and I know there is
nothing you can do to make them change their mind).

2013/12/11 Paul Belanger paul.belan...@polybeacon.com:
 On 13-12-09 06:47 PM, Bryan Anderson wrote:

 I have a call queue that rings about 15 users and they are wanting to set
 it up so that the last person to answer a call doesn't ring on the next
 incoming call.

 What would be the best way to handle this?  I have been looking at the
 strategies and none of those seem to be right for this.  My current
 thoughts are probably a macro that places a penalty on the user tell the
 next call is answered.

 Any advice for this would be greatly appreciated.

 You have agents that log into a queue that don't want to get calls? Is that
 what you are saying?

 Options 1 - log the agent out, they don't get the next call.

 Option 2 - Set up weights for your agents, as answer a new call, increment
 then up so they don't get the next.

 Either way, I see issues with the setup.  Best ways is to rethink your queue
 strategy and stop using ring all.

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Re: [asterisk-users] Calls Recording Solution

[Lenz Emilitri]

We have a number of clients using OrecX and they are quite happy about it.
l.

2013/10/22 bilal ghayyad bilmar...@yahoo.com:
 Hello;

 I am looking for calls recording solution to do recording based on the
 network traffic .. The solution to be competitive and appreciate if it is
 open source .. Any suggested one?

 Regards
 Bilal

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[asterisk-users] QueueWiz - a free call-center simulator tool for Asterisk

[Lenz Emilitri]

Hello all,
next week it's Astricon 10 time, so we thought we'd create something
that the community could like and use for free. It's a pretty
effective tool if you run a call-center or plan to run one.


QueueWiz is the first free web app for interactive, quick and accurate
call center sizing, cost and revenue simulation. Insert your data with
the intuitive interface, measure traffic intensity, expected wait
times, agents' engagement, revenue per call and per agent and even
hourly margins. Save your simulation and share it via email or social
media.

Completely free of charge - no string attached - try it at
http://queuewiz.queuemetrics.com

Have a great day and see you next week.
l.





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Re: [asterisk-users] Queue Management

[Lenz Emilitri]

This should happen automatically - not sure what you want to do.
l.


2013/9/26 akhilesh chand omakhileshch...@gmail.com:
 Dear All,


 I have six different campaign and  5 different agent have login on that
 campaign.Same thing i have done using agi and database,i never use queue
 management on this scenario. Agent can also shuffling  one campaign to
 anther campaign.
 Now i want to do some work with queue.I want to use single queue to managing
 this.

 Eg:
 campaign   Agent Login

 A   a_1,a_3
 (In campaign A 2 agents are login)
 B   a_2,a_1
 (In campaign B 2 agents are login)
 C   a_3,a_1,a_4
 (In campaign C 3 agents are login)
 D   a_4,a_5,a_3
 (In campaign D 3 agents are login)
 E   a_1,a_3,1_2
 (In campaign E 3 agents are login)
 Fa_5,a_4
 (In campaign F 2 agents are login)

 When a call come to campaign A that call goes to agent a_1 or a_3 not goes
 to other campaigns agents.

 Regards
 Akhilesh



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[asterisk-users] Astricon - let's talk call centers?

[Lenz Emilitri]

Hi list,
I know it's a bit OT, but for those who will be at the Astricon, we
are organizing a very informal meeting (maybe in front of a pint or
two) to talk about Asterisk for call-centers. No marketing or anything
- just a way to exchange ideas and meet f2f.

I created a facebook group to organize it - see

https://www.facebook.com/groups/507826572618269/

See you in Atlanta!
l.




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Re: [asterisk-users] Pull call out of queue

[Lenz Emilitri]

You could transfer to a dead-end extension that plays MOH and then
transfer it back somewhere else. Or to a queue with no agents on (if
you are using queues, most likely you are already monitoring queues,
so this may make your workflow easier to live with).
l.

2013/9/6 Todd R. tjrl...@live.com:
 Trying to figure out the best way to pull an active call out of a queue by
 unique id and put it on hold. I don't want to put it on hold on the agent's
 phone but I want it to be pulled away from the agent's phone and into
 Asterisk limbo somewhere.

 Shortly after I want to pull the same call out of limbo and redirect it back
 to either the same agent or another.

 I was thinking about call parking but, I think parking is more than I need
 and it potentially introduces more complications.

 I will be doing this through the manager interface on Asterisk 1.8.x.

 Any ideas, thoughts or help would be greatly appreciated.

 Thanks in advance for any help.

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[asterisk-users] Setting different caller-id for second leg of the Originate

[Lenz Emilitri]

Hello all,
I would like to set a different caller-id for the second leg of a call
when doing an originate.
For example:

Action: Originate
Channel: sip/1234
Context: mycontext
Exten: 1
Priority: 1
Callerid: 123 123
Async: true

This sets the caller-id correctly when dialing sip/1234, but I would
like to set the caller-id for the second leg of the call (the one that
goes to 1@mycontext) to something different. How do I do that? Would
it be enough to change the caller-id as soon as the call is
successfully connected?

Thanks for any pointers,
l.

PS: I Know one can easily do this by editing the dialplan at
1@mycontext, but this is something we cannot do now.

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Re: [asterisk-users] Need input on scalable system design...

[Lenz Emilitri]

Hi Greg,
I am aware of a couple of solutions that come prepackaged and offer
distributed queues for Asterisk. One of them, that seems to work well and
reliably, is the one from Raynet. I am sure there are more. On the other
side, I have seen a number of in-house solutions where you basically have a
daemon polling queues statuses and redirecting calls based on the relative
wait times. Rough but effective, and can be deployed easily.

About recordings, my suggestion would be to use something to offload them
right from the servers, like Oreka. Have a number of large clients using it
and they are quite happy (plus, the guys supporting it are superb).

Just my two cents,
l.




2013/8/27 Gregory Malsack gmals...@coastalacq.com

   Hey All,

 Growing call center. Currently at about 200 call center staff, running
 about 1000 calls per hour. Gearing up to double that. Not too sure that a
 single server will support that growth. So, I'm trying to come up with ways
 to scale the system and still maintain a simplistic design. So I'd like to
 bounce some ideas around.

 Currently I am running on a Dell 1950, dual quad core 2.33ghz xeons, with
 16gb ram, and 2 tce400p cards. This server is managing the full load of the
 company. We are recording all calls, running ivr, queues, cdr, cel, and web
 for reporting. I currently have another 1950 of the exact same
 specifications as a cold spare.

 Here's where you can see drawings of my current connectivity and an
 optional connectivity I'm contemplating...

 http://www.paydaysupportcenter.com/current.pdfhttp://www.linkedin.com/redirect?url=http%3A%2F%2Fwww%2Epaydaysupportcenter%2Ecom%2Fcurrent%2Epdfurlhash=qLsB_t=tracking_anet
 http://www.paydaysupportcenter.com/option.pdfhttp://www.linkedin.com/redirect?url=http%3A%2F%2Fwww%2Epaydaysupportcenter%2Ecom%2Foption%2Epdfurlhash=CJG1_t=tracking_anet

 As you can see I currently have a separate sql server and a separate
 storage server for the call recordings. This is all working fine.

 However, I'm thinking for scalability I should be looking to migrate to a
 configuration similar to the one in option.pdf. Where I have a VOIP gateway
 server that simply relays traffic and possibly can do some load balancing
 or intellegent routing. But nothing more then that, and possibly a second
 one of these online as a hot failover.

 Then have separate sql, storage, (i forgot it in the pic) web, and
 asterisk servers behind that on separate dedicated network. Here's my
 dilemma though, how do I balance the load across multiple machines for
 scalability...

 Since 95% of our calls come into queues, I need to be able to maintain
 queue stats and presence across all of the servers. Thus far, I've got
 everything except the extensions.conf file into the mysql database. I
 thought about setting up 2 servers, 1 for sales, and 1 for customer
 service, then possibly break out each call queue to it's own server as
 things grow. Just not sure if that's the right way to go.

 Then regarding extensions.conf, I've read that it too can be placed in the
 sql database and accessed via switch. however it's resource intense, so now
 I'm thinking of maybe putting that file on the nfs server for all of the
 boxes to read from.

 As for the design of that file, I was kind of thinking of a modular design
 within the file using various goto's and gosubs. Our business model is
 based on affiliates and corporate marketing, so we have a ton of did's that
 follow the same call flow with minor modifications in some variables, as
 well as variations in call flow, and hours of operation. Thus the modular
 design of the call flow. Then the primary inbound context would simply be a
 list of did's pointing to a goto with a list of the variations and
 variables for the did.

 Ok, now that I've melted your brains thoughts?

 Thanks all in advance for the discussion...
 Greg

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Re: [asterisk-users] Kepress while on Queue

[Lenz Emilitri]

Yes it will work. One interesting option here is adding to the MOH an
invitation to exit and leave your number and the CC will call you back.
Helps you smooth the load during peak times, reduces staff and everyone
wins :)
l.



2013/8/27 Gopalakrishnan N gopalakrishnan...@gmail.com

 Hi,

 Will Keypress option will work when am in the queue and hearing MoH?

 Lets say a caller is waiting in queue and while he is hearing MoH, can he
 key in some DTMF and go to some other queue? is that possible?

 Regards

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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Lenz Emilitri]

You need to do this when the call connects. If you can do this within a
couple of seconds, this is usually good enough to be usable (that's what
we do on the QueueMetrics agents pages).
Thanks
l.



2013/8/3 Timothy Smith timotsm...@gmail.com

 Hello Folks,

 I am setting up a call center but we have few agents so one agent is
 able to handle calls of different languages and different queues. For
 the agent to identify the caller, I want a popup to appear as the
 phone starts to ring with the caller's number, language (selected in
 the IVR), Queue (sales, support etc) and any other information (e.g a
 URL with parameters)

 I can send this information either via netcat (to a client such as
 yac) to a Windows PC but the problem is I do not know when the caller
 is about to be connected to the agent, so that I run the command. If I
 wasn't using queues, it would be easy because  I would run the netcat
 command and then dial the user's extension.

 My Question is: Is there a way I can know when the caller is just
 about to be connected to an agent (when the agent's SIP extension
 starts ringing)?

 There are these settings setinterfacevar, setqueueentryvar,
 setqueuevar in queues.conf but when can I use them?

 Have you guys been in this situation before? Any alternative solutions
 (sending caller info to an agent)?

 I am using Asterisk 11 and Windows 7 PCs for agents.

 Thank you!

 Kind Regards,
 Wilson

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Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate

[Lenz Emilitri]

Looks correct to me


2013/6/19 Grant Bagdasarian g...@cm.nl

 Hello,

 ** **

 I’d like to use the AMI interface to originate a call to a context in a
 dialplan, and handoff the dial control to the context.

 ** **

 Whenever I execute the below action, the recipient does ring, but when I
 answer it dials the recipient again. I believe this is because once
 answered the system is going to execute the Context/Exten/Prio in the
 Originate action?

 ** **

 Action: Originate

 Channel: Local/outbound1@originateDialContext

 CallerID: 00311234567

 Context: originateDialContext

 Exten: outbound1

 Priority: 1

 Variable: recipient=0031612345678

 Timeout: 1

 ** **

 [originateDialContext]

 exten = outbound1,1,Wait(1)

 exten = outbound1,n,Set(recipient=${recipient})

 exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)

 ** **

 Anyone have an idea how to fix this?

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Re: [asterisk-users] Queue Limit Callers

[Lenz Emilitri]

You should have different sets of agents logged in to different queues and
you should have a monitor to move them from one queue to the other based on
incoming traffic.
l.


2013/6/17 Shanavaz E A shanava...@yahoo.com

 Hi,

 I have a requirement, which I am not sure whether it can be implemented. I
 had done some searches but didnt find an answer to this. Kindly let me know
 if some one has an idea to implement this:

 I have two Queues - Sales  Booking
 I have 12 Agents who are added to both the queues

 Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales
 Queue.

 Only 8 calls in the Booking Queue should hit the Agents and the other 4
 calls should remain in hold.
 4 calls in the Sales Queue should hit the other 4 agents and the other 2
 call should be in hold.

 Means at a time a maximum of 8 Booking calls only should hit the agents
 and 4 Sales Calls only should hit the agents.

 If number of logged in agents are less, proportionally the number of call
 limit should be reduced. For example, if there are only 10 agents, 7
 Booking Calls should hit and 3 Sales calls should hit. The idea is that all
 agents should be able to answer calls in both queues in rotation. Otherwise
 its possible to add some agents to booking queue and other agents to sales
 queue. But thats not what is required.

 Kindly help if there is some idea to implement this.

 Regards
 Shanavaz.

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Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

[Lenz Emilitri]

Looks yummy! http://phono.com/webrtc


2013/5/31 Adnan 112linuxstockh...@gmail.com

 Voxeo/Phono webrtc.

 /Adnan


 On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.comwrote:


 Hi All,
 I wonder if any of you has some suggestions on which WebRTC
 client/softphone to use for a click-to-dial, webpage hosted solution. Any
 suggestions?
 Thanks
 l.
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[asterisk-users] WebRTC softphone for Asterisk - any suggestion?

[Lenz Emilitri]

Hi All,
I wonder if any of you has some suggestions on which WebRTC
client/softphone to use for a click-to-dial, webpage hosted solution. Any
suggestions?
Thanks
l.
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Re: [asterisk-users] dial and bridge

[Lenz Emilitri]

Thanks all for your help, in the end I was able to do something like:

Action: Originate
Channel: Local/300@from-internal/n
Application: MusicOnHold
Async: 1


As soon as this connects, the callee hears MOH. I get the channel out via
AMI events and start another call:

Action: Originate
Channel: Local/301@from-internal/n
Application: Bridge
Data: Local/300@from-internal-aa8c;1
Async: 1

when this connects, it is immediately bridged to the first callee. I just
have to keep track of errors and hang up the first call if the seconds does
not go through.

Thanks a lot!
l.




2013/5/15 Dan Cropp d...@amtelco.com

 You could use AsyncAGI to achieve this.

 ** **

 Originate the first call (passing in some unique identifier as a
 variable), then using AMI you will see the channel data.  When you see an
 Event: AysncAGI for that channel (with that id, you have control of the
 call).  Send a Dial Action telling it to dial the call and bridge them
 together if the person answers.  If they don’t answer, you will be notified
 and can do something with the original call (play a message, hangup, etc).
 If they are bridged, you can see how long, etc.

 ** **

 Setup an extension, naming it something like patching

 ** **

 exten = patching,1,AGI(agi:async)

 ** **

 Action: Originate
 Channel: Local/300@from-internal

 Async: 1
 Exten: 1

 Context: patching
 Data: 1973

 Variable: YourUniquePatchID=1234

 ** **

 ** **

 Using AsyncAGI and AMI, you can have full control of the call.  You do
 have to setup a very simple dial plan so Asterisk knows you are using
 AsyncAGI to control the call.

 ** **

 Have a great day!

 Dan

 ** **

 ** **

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
 *Sent:* Tuesday, May 14, 2013 11:16 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] dial and bridge

 ** **


 

 Hi all,

 I need some advice - I have been working on originating multiple calls
 using AMI and then joining them. 

 What I want to do is:

 - dial call 1 (where the caller is in a channel format, like SIp/1234 or
 Local/1234@ext) and park it somehow

 - dial call 2 (where again the caller is in channel format) and join it to
 the previous call.

 ** **

 As a requirement, I cannot use the dialplan as an end-point (as I cannot
 change it) but need to use the AMI only.

 ** **

 I tried doing something like:

 ** **

 Action: Originate
 Channel: Local/300@from-internal

 Async: 1
 Application: Wait
 Data: 1973

 ** **

 So that the call goes to 300 and then basically stays there forever, and
 then I dial again:

 ** **

 Action: Originate
 Channel: Local/500@from-internal

 Async: 1
 Application: Wait
 Data: 1973

   

 And then try to bridge the results, but it does not seem to work.

 What I would like to do would be more on the lines of:

 ** **

 Originate call 1 and park it (using a park or waiting)

 Originate call 2 and bridge it immediately to call1 (using the Application
 part)

 ** **

 But maybe I am missing something? is there anybody who has better
 suggestions?

 ** **

 Thanks

 l.

 ** **

 ** **

 ** **

 ** **

 ** **

 ** **

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Re: [asterisk-users] dial and bridge

[Lenz Emilitri]

Hi Mitul,
I agree that the dialplan way is easier, but it's a client requirement to
avoid using it. I was wondering if there was a way to send a call directly
to a parking slot right from the originate, because that is cheaper than
running conferences, and then joining the second call right to the parked
call, so that all we have to do is two originates.
l.


2013/5/14 Mitul Limbani mi...@enterux.in

 Dial first call and put it into a conference, then dial second call and
 put him into same conference to bridge both.

 However dial plan way is much more simpler.

 Mitul


 On Tuesday, May 14, 2013, Lenz Emilitri wrote:


 Hi all,
 I need some advice - I have been working on originating multiple calls
 using AMI and then joining them.
 What I want to do is:
 - dial call 1 (where the caller is in a channel format, like SIp/1234
 or Local/1234@ext) and park it somehow
 - dial call 2 (where again the caller is in channel format) and join it
 to the previous call.

 As a requirement, I cannot use the dialplan as an end-point (as I cannot
 change it) but need to use the AMI only.

 I tried doing something like:

 Action: Originate
 Channel: Local/300@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 So that the call goes to 300 and then basically stays there forever, and
 then I dial again:

 Action: Originate
 Channel: Local/500@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 And then try to bridge the results, but it does not seem to work.
 What I would like to do would be more on the lines of:

 Originate call 1 and park it (using a park or waiting)
 Originate call 2 and bridge it immediately to call1 (using the
 Application part)

 But maybe I am missing something? is there anybody who has better
 suggestions?

 Thanks
 l.






 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com



 --
 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel,
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-71967121
 Cell: +91-9820332422



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 _
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Re: [asterisk-users] dial and bridge

[Lenz Emilitri]

Hi Warren,
the problem is that all I have is two channels, so the specs might be join
SIP/123 and SIP/345 not join SIP/123 to 456@from-internal. They might be
Local channels, but this should be able handle the general case. The reason
why I have channels and not ext@ctxt is that I read them live from the AMI
itself. any idea on how to do this?
Thanks
l.



2013/5/14 Warren Selby wcse...@selbytech.com

 On Tue, May 14, 2013 at 11:16 AM, Lenz Emilitri lenz.lo...@gmail.comwrote:


 Hi all,
 I need some advice - I have been working on originating multiple calls
 using AMI and then joining them.
 What I want to do is:
 - dial call 1 (where the caller is in a channel format, like SIp/1234
 or Local/1234@ext) and park it somehow
 - dial call 2 (where again the caller is in channel format) and join it
 to the previous call.



 Why not just originate from one extension to the other?  Something like
 this (not tested):

 Action: Originate
 Channel: Local/300@from-internal
 Context: from-internal
 Exten: 500
 Timeout: 30

 Should dial extension 500 in the from-internal context after the call to
 300@from-internal is answered.  Meaning, the person at 300@from-internalwould 
 have their phone ring, they'd pick it up, and then they'd hear
 ringing on the line as asterisk then dialed extension 500@from-internal.



 --
 Thanks,
 --Warren Selby, dCAP
 http://www.SelbyTech.com http://www.selbytech.com

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Re: [asterisk-users] dial and bridge

[Lenz Emilitri]

I never actually used parking, but should it work if I call the Park
application as the second leg of the Originate (w/o going through the
dialplan)? I dont seem to be able to make it work.
l.


2013/5/15 Mitul Limbani mi...@enterux.in

 The dial n bridge might work, but there ain't indefinite wait in that
 scenario.
 Direct calls to parking you might try Local(70X@from-internal) but I m
 not sure if this method works reliably.

 The method I mentioned is used by vicidial and it works flawlessly, yes it
 comes with some computing load, however you can try the newer ConfBridge
 app to see if its cheaper.

 Mitul

 On Wednesday, May 15, 2013, Lenz Emilitri wrote:

 Hi Mitul,
 I agree that the dialplan way is easier, but it's a client requirement to
 avoid using it. I was wondering if there was a way to send a call directly
 to a parking slot right from the originate, because that is cheaper than
 running conferences, and then joining the second call right to the parked
 call, so that all we have to do is two originates.
 l.


 2013/5/14 Mitul Limbani mi...@enterux.in

 Dial first call and put it into a conference, then dial second call and
 put him into same conference to bridge both.

 However dial plan way is much more simpler.

 Mitul


 On Tuesday, May 14, 2013, Lenz Emilitri wrote:


 Hi all,
 I need some advice - I have been working on originating multiple calls
 using AMI and then joining them.
 What I want to do is:
 - dial call 1 (where the caller is in a channel format, like SIp/1234
 or Local/1234@ext) and park it somehow
 - dial call 2 (where again the caller is in channel format) and join it
 to the previous call.

 As a requirement, I cannot use the dialplan as an end-point (as I
 cannot change it) but need to use the AMI only.

 I tried doing something like:

 Action: Originate
 Channel: Local/300@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 So that the call goes to 300 and then basically stays there forever,
 and then I dial again:

 Action: Originate
 Channel: Local/500@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 And then try to bridge the results, but it does not seem to work.
 What I would like to do would be more on the lines of:

 Originate call 1 and park it (using a park or waiting)
 Originate call 2 and bridge it immediately to call1 (using the
 Application part)

 But maybe I am missing something? is there anybody who has better
 suggestions?

 Thanks
 l.






 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com



 --
 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel,
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-71967121
 Cell: +91-9820332422



 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 To UNSUBSCRIBE or update options visit:
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 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com



 --
 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel,
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-71967121
 Cell: +91-9820332422



 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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[asterisk-users] dial and bridge

[Lenz Emilitri]

Hi all,
I need some advice - I have been working on originating multiple calls
using AMI and then joining them.
What I want to do is:
- dial call 1 (where the caller is in a channel format, like SIp/1234 or
Local/1234@ext) and park it somehow
- dial call 2 (where again the caller is in channel format) and join it to
the previous call.

As a requirement, I cannot use the dialplan as an end-point (as I cannot
change it) but need to use the AMI only.

I tried doing something like:

Action: Originate
Channel: Local/300@from-internal
Async: 1
Application: Wait
Data: 1973

So that the call goes to 300 and then basically stays there forever, and
then I dial again:

Action: Originate
Channel: Local/500@from-internal
Async: 1
Application: Wait
Data: 1973

And then try to bridge the results, but it does not seem to work.
What I would like to do would be more on the lines of:

Originate call 1 and park it (using a park or waiting)
Originate call 2 and bridge it immediately to call1 (using the Application
part)

But maybe I am missing something? is there anybody who has better
suggestions?

Thanks
l.






-- 
Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com
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[asterisk-users] amiDebugger - might make your life easier if you program through the AMI

[Lenz Emilitri]

Hi all,
I have been playing with the AMI quite a bit lately - mostly debugging
WombatDialer in production, but that's a different story - and I have been
frustrated by the lack of a simple way to interact CLI-like with the AMI
itself. So I have decided to write something myself to make my life easier,
or at least a bit less miserable.

The result is a little webapp that you can use as a sort of CLI-frontend to
the AMI itself. It is not pretty, but pretty much effective. So I thought I
could share it and make someone else's life a bit easier.

You can find it on https://github.com/l3nz/amiDebugger  - if you just want
to test-drive it get the WAR file an put it into some webapp container,
e.g. Tomcat.

Hope you'll like it.
l.


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Test-drive WombatDialer beta @ http://wombatdialer.com
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Re: [asterisk-users] looking for a way to do appointment reminders

[Lenz Emilitri]

We did something like that - see
http://blog.wombatdialer.com/post/24187267017/drstrangelove
You can use the free version of the dialer if you have low traffic or just
want to run a test.
l.


2013/4/26 Ron Wheeler rwhee...@artifact-software.com

 Good comment.
 Another feature suggestion
 You might to ask the person to press 1 to confirm or 2 to leave a message
 if the appointment is not going to be kept or 0 to reach the receptionist
 to reschedule the appointment.

 Ron


 On 26/04/2013 7:06 AM, Chris Bagnall wrote:

 On 26/4/13 10:38 am, jg wrote:

 they are currently calling patients. I think these calls apply only to a
 certain fraction of the patients, who are difficult to contact by other
 methods.


 I suspect there will be different requirements depending on how 'helpful'
 to patients you wish to be. At the very simplest end of the scale, you
 could simply call the patient's number and remind them of their appointment
 on dd hhmm, then disconnect.

 However, the OP probably wants something a little more sophisticated than
 that. At the very least, you would want some method of handling shared
 numbers (e.g. a shared dwelling with a single phone), so you didn't
 inadvertently advertise a patient's appointment to someone else who
 answered the phone. So you would at the very minimum want a simple IVR that
 says We are trying to reach Mr. Joe Bloggs. If this is he, press 1 now,
 otherwise please hang up.

 Going beyond that, you might want your reception staff, when booking
 appointments, to ask the patient when they would like their reminder call -
 the day before, an hour before, etc. etc. (and if the day before, would
 they prefer it in the morning, afternoon, or evening).

 As others have said, the OP might be best advised to request (paid)
 assistance with the project on the [asterisk-biz] list.

 Kind regards,

 Chris



 --
 Ron Wheeler
 President
 Artifact Software Inc
 email: rwhee...@artifact-software.com
 skype: ronaldmwheeler
 phone: 866-970-2435, ext 102



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Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger

[Lenz Emilitri]

Depends on what you are trying to do. Not in general (AFAIK) but you may
find a number of scripts around.



2013/4/18 isr...@gmail.com

 I think facebook uses xmpp so you could use asterisk jabber or so
 Don't know about the rest

 -Original Message-
 From: bilal ghayyad bilmar...@yahoo.com
 Sender: asterisk-users-boun...@lists.digium.com
 Date: Wed, 17 Apr 2013 14:41:53
 To: asterisk-users@lists.digium.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber,
 yahoo and hotmail messanger

 Hello;

 Is there any modules or channels or integration between asterisk and any
 of the following:

 whatsapp, facebook, viber, yahoo and hotmail messanger?

 Regards
 Bilal

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Re: [asterisk-users] Phpagi action based on outbound call user response

[Lenz Emilitri]

I am not sure about PHP AGI, but in general via AGI you can monitor the
state of the call and so you can know when the call is over.
l.


2013/4/17 Rahul R rahul...@gmail.com

 Hello List,

 In PHPAGI, I'm using the Astrisk Manager function send_request() to
 originate an outbound call. I want to execute the remaining PHP code after
 the call gets executed (depending on user input). But presently the call
 originates in a different context and asterisk executes the remaining code
 in parallel.
 Is there a way in which I can pause the code execution until the call is
 completed.

 Note: I wish to return to the context from which the call was originated
 and continue execution.

 Any help is greatly appreciated.
 --
 Thanks  Regards
 Rahul
 http://about.me/rahulr92
 +919567607741

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Re: [asterisk-users] External call control for Asterisk

[Lenz Emilitri]

Not sure if that's what you are looking for, but I would think about having
the dialplan call a web service (maybe using CURL) and passing account and
current number. The system would reply with the number to actually dial, or
none if blocked, and the maximum possible call length. Then it's all
Asterisk (or turtles all the way down).


2013/4/10 Simon Green simon.c.gr...@gmail.com

 Hi there, I’m new to Asterisk and there’s a ton of documentation. I’m not
 really sure where to start. What I want to do is this: a PBX service ala
 FreePBX, but where call control is passed via SIP to an external service
 which will tell Asterisk:



 a)  * Whether the call is allowed

 b)  * Where to connect the call, if necessary (i.e. forced
 redirection to a C-party)

 c)   * To disconnect the call at some time in future based on
 charging considerations (i.e. online charging)



 There is also the option of not using Asterisk at all, and simply using
 the other service directly, but Asterisk is much better suited to handling
 end-user devices. The external service does control logic only.


Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com
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Re: [asterisk-users] 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger

[Lenz Emilitri]

I'd start from
https://github.com/venomous0x/WhatsAPI/blob/master/README.mdthat
offerts PHP and Java APIS, both not hard to integrate with Asterisk.



2013/4/19 kingman chui chuiking...@yahoo.com.hk

 Hi,
   So , how to connect asterisk to whatapps ??Please advice ..
 Thank
 Regard/chui king man

*寄件人︰* Lenz Emilitri lenz.lo...@gmail.com
 *收件人︰* isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial
 Discussion asterisk-users@lists.digium.com
 *傳送日期︰* 2013年04月19日 (週五) 4:34 PM
 *主題︰* Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo
 and hotmail messanger

 Depends on what you are trying to do. Not in general (AFAIK) but you may
 find a number of scripts around.



 2013/4/18 isr...@gmail.com

 I think facebook uses xmpp so you could use asterisk jabber or so
 Don't know about the rest

 -Original Message-
 From: bilal ghayyad bilmar...@yahoo.com
 Sender: asterisk-users-boun...@lists.digium.com
 Date: Wed, 17 Apr 2013 14:41:53
 To: asterisk-users@lists.digium.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber,
 yahoo and hotmail messanger

 Hello;

 Is there any modules or channels or integration between asterisk and any
 of the following:

 whatsapp, facebook, viber, yahoo and hotmail messanger?

 Regards
 Bilal

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com/ --

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Re: [asterisk-users] ACD problem

[Lenz Emilitri]

I am not sure I understand the required routing pattern, but I'm sure
queues are your friends, as you can dynamically add and remove member and
you can have a first-level queue easily move fall-through to another queue
in case all members should be busy or none should be available. Plus by
using queues you decouple the what you want to do from the who is doing
it.


2013/4/10 Tommy Cooper tomcoope...@yahoo.com

   Hi,

 I am working on a small inbound call center solution that uses an ACD
 system. I might add an IVR system later on. I only have 2 extensions set up
 (extensions 1000 and 1001), I want the system to put new calls in a queue
 if both extensions are busy. I am currently subscribed with a SIP trunk
 provider and can successfully recieve calls. I want to design a system
 where customers can call my number, that call will then be directed to
 either extension 1000 or 1001. If both extensions are in use, I want that
 3rd call to be queued.
 I don't think that the config below will direct calls to extension 1001
 because the second line states that any incomming calls should be routed to
 extension 1000. How do I change this so that calls are directed to all of
 my exensions?

 extensions.conf
 [from-myprovider]
 exten = *DID number*,1,Answer
 exten = *DID number*,2,Dial(SIP/1000)
 exten = *DID number*,3,Queue(support) ;not sure if this line belongs here
 exten = *DID number*,4,Hangup

 queues.conf

 [general]
 [support]

 musicclass=default
 strategy=rrmemory
 joinempty=no
 leavewhenempty=yes
 ringinuse=no
 Member = SIP/1000
 Member = SIP/1001

 agent = 1000,1000
 agent = 1001,1001

 When using the current config the caller will listen to the 'music on
 hold' until the agent answers but calls are only being forwarded to
 extension 1000 as stated above

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Re: [asterisk-users] To queue or not to queue...

[Lenz Emilitri]

Hello Gregory,
I wouldn't say this is a typical scenario for using a ringall queue,
especially if the agent set gets larger and larger. On the other side, a
ringgroup won't solve the issue of ringing all those phones at once.  What
I would be looking into, considered the motivation of your agents, is to
split the system into more than one queue and send the calls randomly to
each queue. If everybody is busy you get out and retry. This should not
impact call answer times as long as you have 30/40 people available per
queue - but your box will handle a fraction of the load and you can easily
partition such a system on multiple boxes.
Just my two cents,
l.


2013/3/28 Gregory Malsack gmals...@coastalacq.com

 Hello All,

 History ~
 I recently took a position with a call center. At the time they had about
 50 agents in a call queue. The queue was setup to ringall. The agents use
 Eyebeam softphones. Everything is local lan, no routers, everything
 connected via Cisco 3600 10/100 switches.

 Now we are up to about 150 agents, and I have kept everything pretty much
 the same way for a couple of reasons. However, those reasons are slowly
 drifting away and it's become the right time for me to start questioning
 some of the previous configuration.

 Here's the scenario~
 150 agents, all are commission based sales reps. 99% of the calls are
 answered within the first ring. the rest are answered between the second
 and third ring. Never in my 4 months with the company has a queue call been
 in the queue more then 20 seconds.

 Problem~
 Several times a week or sometimes a day, the reps will tell me that the
 same call will be answered by 3 or 4 or 5 reps, and none of them get the
 inbound audio. Asterisk only shows 1 of the reps actually connecting the
 call, however the call logs in Eyebeam for all 5 reps, show that they took
 the call and were connected for a short period of time before disconnecting
 the call because there is no inbound audio.

 Point of discussion~
 Is there really a reason to maintain a queue? With the companies growth
 they are now discussing the option of sending certain affiliates to certain
 sales reps. Am I better off using ring groups? Additionally I am working
 towards running as much of my configs via mysql as possible and turning up
 multiple servers to handle the calls. So far we have reached 130
 simultaneous calls on one server, and about 10,000 calls processed during a
 12 hour day.

 Thanks for reading. I look forward to hearing peoples views on this...


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Re: [asterisk-users] Dedicated LCR Solutions

[Lenz Emilitri]

I know Evariste Systems has a product called CSRP -
http://evaristesys.com/pub/CSRP-ProductOverviewCapabilitiesSurvey.pdf -
that looks very interesting and it is built for high-volume scenarios. It
is basically a standalone box you route calls to.
Just my two cents,
l.



2013/3/26 Nick Khamis sym...@gmail.com

 Hello Everyone,

 Was wondering what some of you for stand alone LCR implementations. I
 am aware of the LCR module within asterisk and a2billing however, we
 are looking for a standalone self less coupled solution. Not sure if
 such thing exist. Kind of like CDR Tool but for LCR...

 Thanks in Advance,

 Nick

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Re: [asterisk-users] Configuration Required for Remove Queue Member

[Lenz Emilitri]

Sounds like the autopause option?
l.


2013/1/28 Ahmed Munir ahmedmunir...@gmail.com

 I would like to know, is there a method in which  we can define the
 timeout value for a member who already login to the queue but after quite a
 while if he didn't answer the 3-4 calls (not going to member pause queue)
 but automatically remove the member from the queue?

 Please advise.



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Re: [asterisk-users] Integration with Social Media, Email and Web call center

[Lenz Emilitri]

I was thinking of something similar, maybe using the URL field of the
queue() app as to point to an internal broker that will then link to the
message being used.
In theory one could do this for all kinds of traffic, including e-mails.
The part I don't really like is keeping an audio call open for the duration
of the job, but it plays very well with existing queues.
In the end, I guess Matt is always right :)
l.

2013/1/24 Matt Riddell li...@venturevoip.com


 In the past I've sent calls to an agent in the queue with music on hold
 that contained a beep every 20 seconds (to remind them they're on a call)
 and then used the same code I do for screen popping to send them
 alternative records.  I.E. web page, email, fax etc.  It's stored in the
 database that that's what they were working on and then when they finished
 working on it they just hang up or press * to disconnect the call.

 That way you can use the standard Asterisk queues and they don't get
 bothered by anything else while they're working on it.

 Facebook might be a little harder as you wouldn't necessarily know when an
 incoming request came.

 --
 Cheers,

 Matt Riddell
 ___

 http://www.venturevoip.com/news.php (Daily Asterisk News)
 http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk)
 http://www.venturevoip.com/exchange.php (Full ITSP Solution)
 http://www.venturevoip.com/cc.php (Call Centre Solutions)






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Re: [asterisk-users] Capture queue agent drop and put caller back in queue

[Lenz Emilitri]

2013/1/21 Mitch Claborn mitch...@claborn.net

 Asterisk 11

 Occasionally we will have a partial power outage, or a piece of network
 equipment will fail, and our queue agents who are on active calls with
 callers will be disconnected from the caller.  What I'd like to do is
 capture those calls and put them back in the queue (at a high priority) so
 that we don't lose the caller.

 I've tried to duplicate the situation in my lab: I have one agent in the
 queue, a caller dials into the queue, gets connected to the agent then I
 pull the ethernet cable out of the agent's computer (testing with a
 softphone) but I don't see anything happen on the asterisk console.  core
 show channels shows the 2 channels still bridged even though the agent is
 gone.

 Shouldn't asterisk somehow know when the agent disappears?
 How can I accomplish my goal?





I am not sure that from the PoV of the caller this solution would work -
they would experience tens of seconds of silence plus they would have to go
back to the queue. If this happens rarely, you could have a process call
them back instead - you acknowledge what happened and have someone on-line
with the person apologizing.

We have a few clients implementing something like this for calls exiting
the queue on timeouts and it seems to be well-liked by the callers. Of
course it depends on what you are doing and the level of service that
callers come to expect.

Just my two cents,
l.




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Re: [asterisk-users] Integration with Social Media, Email and Web call center

[Lenz Emilitri]

And how would you have this working together with Asterisk queueing? I have
seen solutions like this using agent pauses and then making everyithing
happen outside the normal ACD flow, but it's a bit of a hack
l.


2013/1/22 Danny Nicholas da...@debsinc.com

 For just the messaging part, you should be able to use wget or curl to
 interface and create messages.  You might have to go a little higher
 level
 like C or Perl, but it sounds very doable.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal
 ghayyad
 Sent: Tuesday, January 22, 2013 4:27 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Integration with Social Media, Email and Web call
 center

 Dears;

 Can someone advise me where to find a technology (open source) that let us
 able to integrate with social media like whatsapp and facebook? And use
 this
 in call center (queuing the messages and routing it for agent)?

 Anyone give me a light to start?

 Regards
 Bilal



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Re: [asterisk-users] Detect Low Quality Calls - Realtime

[Lenz Emilitri]

2013/1/5 joachim zoach...@securax.org


 You are pretty much limited to measuring the delay and the jitter.
 The delay you can somewhat estimate prior to the call (with qualify for
 example).
 The jitter / packetloss you can only figure out when the call is already
 up for a while. (e.g. you might have no issues the first minute, but maybe
 packet loss will come in bursts after a minute).


A few years ago I spoke to a Finnish company that had a commercial solution
for automated MOS estimation. So something exists though I have not tested
it first-hand.
l.

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Re: [asterisk-users] new user help required to build voice recorder with asterisk

[Lenz Emilitri]

I don't think this should be an issue, but we have seen a lot of sites
going live and discovering too late that they had recording problems. Maybe
you won't need to implement an external recorder, but it's better to plan
in advance, not when you are in production! :)
l.


2013/1/2 Leandro Dardini ldard...@gmail.com

 I don't know how many I/O can be achieved on a modern hardware, but I
 don't think 60 concurrent calls will be a problem. 60 calls are just 4
 Mbit/s of data. However can be a good idea to start loading a server and be
 prepared to share the load on another server.

 Leandro



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Re: [asterisk-users] Paging for Praying

[Lenz Emilitri]

How many people do you plan to page? because if numbers are high (or
variable) you may have an easier life by using some sort of dialer if
numbers are not very high and two lines are enough, our WombatDialer is
free to use.
l.


2012/12/29 bilal ghayyad bilmar...@yahoo.com


 2) Praying time need to be obtained from text (or database). So, it is not
 always the same time. What actually is needed to be obtained from the text
 file or the database is the time of the pray for each date (for example, if
 today is 28 of December so the query will be for this date and then it is
 required to check if the time is same as the current time to page the wave
 file on the Phones).



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Re: [asterisk-users] new user help required to build voice recorder with asterisk

[Lenz Emilitri]

With just one PRI card this should not be an issue, but for larger systems
you may consider using something like Oreka to offload the I/O from the
Asterisk server
l.


2012/12/31 Vinod Nadiadwala thinw...@gmail.com

 Hi,

 I am new to asterisk, i want to know that is it possible to use asterisk
 for build voice recording system.

 Scenario :
 ISDN PRI line (30 line)
 I want every incoming  outgoing call has to recorded, but without manual
 action. system has to auto receive the call.

 Please suggest, how should i start and with which hardware / cards it is
 possible.




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Re: [asterisk-users] Catching hold in dialplan

[Lenz Emilitri]

Steve Murphy submitted a patch a while ago to track MOH on queues, you can
find it at https://issues.asterisk.org/jira/browse/ASTERISK-20742 - it
could be a good starting point to work on as it is quite short.
Too bad it is still in limbo :-(
l.



2012/12/19 Andrew White and...@computersforall.com.au

  Hey all,



 I’ve built a custom application for our call center and am having one
 problem. Unfortunately certain things happen whilst the agent has the
 customer on hold which I’d like to work around. But I can’t work out how to
 catch the actual hold event so I can do something about it. From the
 console with verbosity on 12, all I can see is:

 -- Started music on hold, class 'default', on SIP/trunk-9546

 -- Started music on hold, class 'default', on SIP/100-9547



 I’m happy to try and catch this AGI or via manager if needed, however a
 dialplan based solution would be best.



 Thanks all!



 Andrew

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Re: [asterisk-users] Users list email totals by year .

[Lenz Emilitri]

So where has every body else gone? :)
l.


2012/12/30 Mr. James W. Laferriere bab...@baby-dragons.com


 2003, 24471
 2004, 48608
 2005, 59116
 2006, 41215
 2007, 26414
 2008, 20746
 2009, 18304
 2010, 14948
 2011, 11588
 2012, 7542

 --
 +--+
 | James   W.   Laferriere | SystemTechniques | Give me VMS |
 | NetworkSystem Engineer | 3237 Holden Road |  Give me Linux  |
 | bab...@baby-dragons.com | Fairbanks, AK. 99709 |   only  on  AXP |
 +--+

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Re: [asterisk-users] It's possible a redudant Queue?

[Lenz Emilitri]

We have a number of customers who use this approach with local or
geographically distributed Asterisks and then use QM clustering to observe
the system as if it was one single big box. Seems to work fine and it' easy
to set up and maintain.
l.



2012/12/14 Danny Nicholas da...@debsinc.com

 In my experience, you should set up two identical queues and
 configurations.
 With a little work, you should be able to let server 1 know the phone is in
 use by server 2 and vice versa.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo
 Dionisi
 Sent: Friday, December 14, 2012 9:49 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] It's possible a redudant Queue?

 Hi all,
 I have a doubt. I have to create a queue with 3 phones, these phones can be
 reached via two redudant Asterisk server.

 I can pass a variable (the sip trunks) to the queue or should I do two
 queues with the different trunks?

 Danilo



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Re: [asterisk-users] Queue logging

[Lenz Emilitri]

How large is your systems? because the information created by of a call on
a queue is just like a hundred bytes, so it is usually safe to keep them
all in any case on modern systems.


2012/11/27 Jonas Kellens jonas.kell...@telenet.be

  Hello,

 at the moment I am logging queues into a MySQL DB, but this can quickly
 become a lot of information.

 Is there a way to exclude certain queues from being logged into the queue
 log ?



 Thanks,
 Jonas.

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Re: [asterisk-users] Queue_log into MySQL - best practices

[Lenz Emilitri]

Hi Dmitry,
we usually advise against writing queue_log events straight to a database,
as it is marginally more likely that the DB has issues that a simple flat
file. And when data is lost it's lost forever. Still everybody seems to
love writing data straight to the DB :)
l.


2012/11/22 Dmitry mbike200...@yahoo.com

 Hi,

 I use asterisk 1.8.
  Currently I use a perl daemon to parse queue_log into MySQL. It works
 reliably.

 But I know that there is a method (
 http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL and
 http://work.mikeboylan.com/asterisk-queuelog-to-mysql) to write to MySQL
 directly with app_mysql which has a DEPRECATED status.

 My question is:
 What is the best/preffered approach to put queue_log into MySQL in
 asterisk 1.8 and up?
 1) To use external daemons to parse /var/log/queue_log?
 2) To use the deprecated app_mysql? the status does not guarantee that
 this application will be in the future
 3) To use odbc to access mysql? but I could not find a procedure for it.
 And I doubt it is possible.

 BR,
 Dmitry Pavlenko




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Re: [asterisk-users] Astricon 2012 presentations

[Lenz Emilitri]

Thanks - too bad I missed it :)


2012/11/12 Dan Jenkins dan.jenk...@holidayextras.com

 Hi,

 As far as I'm aware the videos are still being produced and there's no
 definitive list anywhere for the slide decks.

 However, my one is here:
 http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614

 Dan Jenkins

 --
 Dan Jenkins - Senior Web Developer
 email: dan.jenk...@holidayextras.com
 twitter: dan_jenkins http://twitter.com/dan_jenkins
 linkedin: jenkinsdaniel http://www.linkedin.com/in/jenkinsdaniel
 skype: d-jenkins
 blog: www.dan-jenkins.co.uk
 about.me: about.me/dan_jenkins



 On 12 November 2012 11:05, Lenz Emilitri lenz.lo...@gmail.com wrote:

 Hello all,
 anybody knows if the PDFs for presentations held at Astricon 2012 are
 available somewhere? I looked at the website but cannot find anything.
 Thanks
 l.


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Re: [asterisk-users] Asterisk 1.8.16 Monitoring tools

[Lenz Emilitri]

Hello Motty,
it really depends on what you want to do and the level of detail you want.
There are a number of free and commercial applications that can help you in
doing this :)
l.


2012/11/9 motty.cruz motty.c...@gmail.com

 Hello,
 I want to monitor my Asterisk 1.8, inbound, outbound, status calls, queue
 call? Any suggestions?

 I found Monast, I'm having issues configurating.

 Thanks,


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[asterisk-users] Astricon 2012 presentations

[Lenz Emilitri]

Hello all,
anybody knows if the PDFs for presentations held at Astricon 2012 are
available somewhere? I looked at the website but cannot find anything.
Thanks
l.


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Re: [asterisk-users] how to lookup a call

[Lenz Emilitri]

I would not know if this is something that can be helpful to you, but in
WombatDialer we associate a channel variable to with an unique-id to each
call, so that we can reattach to a set of calls if the AMI connection goes
down and we can be absolutely sure that what we are looking at is the call
we think it is. It is not really expensive to do - just a GetVar per
channel to mek sure our assumptions are correct.


2012/11/7 Jerry Geis ge...@pagestation.com

 I am using 1.4.43 currently.

 I am using the AMI to originate a call over a SIP Trunk to my cell
 XXX506. works fine.
 when the call is active I do a core show channels concise and I get:

 SIP/testsystem-0ad0!**smvoice-dialout!callprogress!**
 4!Up!AGI!smvoice!0!!3!24!(**None)

 My AGI is called smvoice.
 No place does my number show up.
 How do I lookup my call so I can hangup the call at a later time.

 In my case there my be more than one call active at a time, and I want to
 hangup the correct call. I know I need the data testsystem-0ad0 to
 cancel my call
 but how do I associate that with my number so I can find the right call
 to hangup.

 Thanks,

 Jerry

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Re: [asterisk-users] Agents in more than one queue at once

[Lenz Emilitri]

In general there is no guaarantee as which call will connect; each queue is
independent AFAIK.
l.


2012/10/17 Alex Forster a...@alexforster.com

 My company has been running Asterisk 1.6.2.19-1_centos5 from the official
 yum repo, and for a while now I've been receiving complaints from our call
 centers about calls not being routed in the most efficient order.

 I'll explain with a simplified scenario--

 Let's say I have two queues: A and B. I have one agent, Alice, who is a
 member of both of these queues. While Alice is busy on a call, one person
 calls in to queue A, and then, several moments later, another person calls
 in to queue B.

 At this point, note that both callers waiting on hold are position 1 in
 their respective queues. A queue show might look like this...

  A has 1 calls (max unlimited) in 'leastrecent' strategy (0s
 holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s
 Members:
21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls
 (last was 533 secs ago)
 Callers:
1. SIP/Trunk-eb17 (wait: 1:14, prio: 0)
 
  B has 1 calls (max unlimited) in 'leastrecent' strategy (0s
 holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s
 Members:
21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls
 (last was 533 secs ago)
 Callers:
1. SIP/Trunk-eb1e (wait: 0:45, prio: 0)

 My question is: when Alice gets off the phone, which call will she get? My
 expectation is that she will get the call which has been waiting longer,
 but I'm not sure that's actually the case.

 Alex Forster




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Re: [asterisk-users] Call me now outbound calls in a queue

[Lenz Emilitri]

The problem is that you need to have a process waiting for a free agent and
then doing the reschedule. Instead of writing your own, you could try our
WombatDialer (that is currently free as in beer, as it is being community
tested) to automate such a  task. It has a nice HTTP API and it would do
exactly what you are looking for.
See http://wombatdialer.com/
l.

2012/9/28 Mitch Claborn mitch...@claborn.net

 That approach only works if there are any agents that are not busy on a
 call - I could pick one, take them out of the queue then connect the call.
  If all agents are busy, I need to be able to insert the request into the
 queue so that it gets processed in sequence with the inbound calls.




 Mitch



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Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/

[Lenz Emilitri]

The problem I see with this approach is that you usually do not just want
to dial out 10 calls at a time, but you will want to keep track of what
happened to them and (in case) reschedule them. So  you will likely need to
monitor them over AMI to make sure they went through, and you need to
implement some rescheduling logic.

[Shameless plug starts here]
This was the reason why we started working on Wombat a while ago - to offer
something that would handle all this (and more) but leaving you the
Asterisk touch of being free to program the call handling at the dialplan
level, so you would get the best of both worlds. Did I already mention the
current beta versions are free? :)
[Shameless plug ends here]

I am not saying that this is the only correct solution (or it is a correct
solution at all) but our almost ten years of Asterisk call-center
experience show that what starts out as something quick and simple to plug
a hole ends up being a platform :)

Just my two Swiss cents,
l.


2012/9/28 A J Stiles asterisk_l...@earthshod.co.uk

 On Friday 28 September 2012, Patrick Archibald wrote:
  Hi,
 
  Is there a way to move 100 .call files in to
  /var/spool/asterisk/outgoing/ at once and have Asterisk call at
  maximum 10 at a time?

 Yes:  Move them in batches of 10.  Could be as simple as
 last if ++$n_files  9;
 if the script is in Perl.

 You know how many calls you can deal with at once; it's up to you to stay
 within your own limits.  Asterisk just tries its damnedest to do whatever
 it's
 been told, without imposing any sort of judgement as to whether it's sane
 or
 wholesome.



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Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

[Lenz Emilitri]

Another option that seems to be very good for handling logs where you write
quite a lot is Cassandra - http://cassandra.apache.org/ - but of course you
lose the SQL layer on top - unless you go for something like
http://blog.mariadb.org/announcing-the-cassandra-storage-engine/

This may not be completely off topic here because you get high data
security / crash protection and parallel cluster writes, so you could
insert tens/hundreds of thousands of events per second on a suitably
dimensioned cluster for an Asterisk server cluster of similar size :)
l.


2012/9/28 Leif Madsen leif.mad...@asteriskdocs.org

 On 27/09/12 11:45 AM, Matt Hamilton wrote:


 Date: Thu, 27 Sep 2012 10:23:35 +0200
 From: lenz.lo...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

 I'd go for MyISAM and would set up a remote replica if data integrity is
 important.

 If you have like 1000 calls of (say) 30 seconds avg length, and you
 create 10 events per call, you would expect an event every three seconds.
 This is about 300 inserts per second. Say 600 at peaks. This should be
 feasible with server-grade hardware without much difficulty. Also as you
 always INSERT it behaves as a log file (no seeking, no locking) if the
 table is optimized.
 l.


 We decided to go with MyISAM since it supports concurrent
  inserts (as you suggested). Data integrity (a slight loss of
 call records) is something we can live by. Right now we use DRBD for
 replication, but I guess with MyISAM it doesn't make much sense if the db
 crashes. We are looking into other options as well.


 This may or may not be relevant, but you can also check out
 MySQL/Galera[0] for clustering solutions. Not sure if that gets you closer
 or further from your goal though :)  It uses a modified InnoDB to allow a
 multi-master MySQL cluster.

 I used a chef cookbook to deploy it[1].

 [0] http://www.codership.com/content/using-galera-cluster
 [1]
 http://support.severalnines.com/entries/21453521-opscode-s-chef-mysql-galera-and-clustercontrol


 --
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Re: [asterisk-users] QUEUEHOLDTIME always zero

[Lenz Emilitri]

What do you get if you run a queue show sales?
l.



2012/9/26 Mitch Claborn mitch...@claborn.net

 Asterisk 1.8.10.1~dfsg-1ubuntu1

 Trying to build a simple announcement of the queue status. QUEUEHOLDTIME
 is always zero.  What am I doing wrong?

 queues.conf
 [general]
 autofill=yes
 shared_lastcall=yes

 [StandardQueue](!)
 musicclass=default
 strategy=rrmemory
 joinempty=no
 leavewhenempty=yes
 ringinuse=no
 announce-frequency = 30
 min-announce-frequency = 15
 announce-holdtime = yes|no|once
 announce-position = limit
 announce-position-limit = 5
 announce-round-seconds = 10
 setinterfacevar = yes
 setqueueentryvar = yes
 setqueuevar = yes

 [sales](StandardQueue) ; create the sales queue using the parameters in
 the StandardQueue template

 extensions.conf
 exten = 812,1,NoOp(queue status)
   same =n,Set(LOGGEDIN=${QUEUE_MEMBER(sales,logged)})
   same =n,Set(READY=${QUEUE_MEMBER(sales,ready)})
   same =n,Set(WAITING=${QUEUE_WAITING_COUNT(sales)})
   same =n,Set(STUFF=${QUEUE_VARIABLES(sales)})
   same =n,Verbose(waiting: ${WAITING} calls in queue: ${QUEUECALLS} avg
 hold: ${QUEUEHOLDTIME} logged in: ${LOGGEDIN} ready: ${READY})

 Regardless of how long a caller has been waiting in the queue, the output
 is:

 -- Executing [812@LocalSets:1] NoOp(SIP/08000F3BE07C-0048,
 queue status) in new stack
 -- Executing [812@LocalSets:2] Set(SIP/08000F3BE07C-0048,
 LOGGEDIN=1) in new stack
 -- Executing [812@LocalSets:3] Set(SIP/08000F3BE07C-0048,
 READY=1) in new stack
 -- Executing [812@LocalSets:4] Set(SIP/08000F3BE07C-0048,
 WAITING=1) in new stack
 -- Executing [812@LocalSets:5] Set(SIP/08000F3BE07C-0048,
 STUFF=0) in new stack
 -- Executing [812@LocalSets:6] Verbose(SIP/08000F3BE07C-0048,
 waiting: 1 calls in queue: 1 avg hold: 0 logged in: 1 ready: 1) in new
 stack
 waiting: 1 calls in queue: 1 avg hold: 0 logged in: 1 ready: 1






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Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

[Lenz Emilitri]

I'd go for MyISAM and would set up a remote replica if data integrity is
important.

If you have like 1000 calls of (say) 30 seconds avg length, and you create
10 events per call, you would expect an event every three seconds. This is
about 300 inserts per second. Say 600 at peaks. This should be feasible
with server-grade hardware without much difficulty. Also as you always
INSERT it behaves as a log file (no seeking, no locking) if the table is
optimized.
l.


2012/9/26 Matt Hamilton mistral9...@hotmail.com

 Our top priority is the raw Write (INSERT) performance, Read (SELECT)
 performance is not important. Strict ACID compliance is not necessary
 either. MySQL (on a separate database server) should be able to handle
 inserting CDR records (approximately up to 10 records for each call) for
 about 1000 concurrent calls coming from an Asterisk cluster.

 Matt



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Re: [asterisk-users] Help Required IVR

[Lenz Emilitri]

Are you programming the dialplan yourself or are you using a GUI? Also, are
you sure that your greetings message is playable by Asterisk?
l.


2012/9/24 Farooq Hussain farooqhussain...@gmail.com

 Hello everyone,

 I stuck in problem I have creating a time based IVR and its working fine.
 If my IVR playing in office hour it would standard IVR and if not they we
 have play a greeting message and place that call to voice mail of
 a extension.

 My problem is this I am able to transfer the call on voice mail but how to
 play greeting message first. I am using trixbox 2.2.8 anyone help is this
 regard would great full.

 --
 Thanks

 Farooq Hussain

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Re: [asterisk-users] BLF and Call Queues

[Lenz Emilitri]

In general I would not use this for a true call-center with hundreds of
agents, where it is the ACD's responsibility to route calls to agents and
there are strict policies on agent behavior, but I'm sure there are a
number of cases where this could be useful (eg small call centers, internal
service desks, receptionists, etc...).
Just my two cents,
l.



2012/8/21 Olivier oza_4...@yahoo.fr

 Hi,

 What about Queue logs ? How is a picked-up call logged ?

 Giving agents the capability to easily pickup a call, without beeing
 logged-in, is a big change with both positive and negative side effects.
 I would be curious to read opinions about that.


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Re: [asterisk-users] Need queue name in CDR

[Lenz Emilitri]

It would likely be easier for you to use a tool that already processes
queue_log information. There are a number available :)

2012/6/13 Pratik Shrestha pratik...@gmail.com

 Dear All,

 I am making asterisk report using CDR values given by asterisk.

 I have queues which consist of multiple members (extension). Also, an
 extension may be in multiple queues. So, I want CDR to record the
 name/number of queue from which the call was originated.


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Re: [asterisk-users] Asterisk Capacity

[Lenz Emilitri]

We are working on a project to create a general-purpose telecasting server
- see http://wombatdialer.com - there is practically no documentation yet,
but it's easy to set up and we tested it originating hundreds of channels
on multiple servers. It is alpha stage, but current versions are free and I
expect them to basically work.

If you want to give it a shot, you can install via RPM as described on the
website.
Thanks
l.


2012/5/3 Ashish Agarwal ashisha...@gmail.com

 So what is a better approach to achieve this
 On May 3, 2012 9:20 PM, Mitul Limbani mi...@enterux.in wrote:

 The other 70 will result into failure with .call file approach.

 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel,
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-61447605
 Cell: +91-9820332422




 On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal ashisha...@gmail.comwrote:

 Hello,

 We are currently working on a project where using .call file on asterisk
 spool, outbound calls will be made from a pri line and a voice clip will be
 played.

 We know that pri has a capacity of handling only 30 channels at a time.
 Therefore, my worry is what happens if we write 100 files at a time on the
 spool. Will asterisk manage the queue or how exactly will it behave.

 Regards,

 Ashish

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Re: [asterisk-users] Asterisk Capacity

[Lenz Emilitri]

At the moment it's free as in beer, though closed-source.
It is written in Java and uses MySQL as its back-end.
l.


2012/5/11 Arstan arst...@gmail.com

 Hi,
 wombat looks promising.

 Questions: What technologies are used? Is it open source license?


 On Fri, May 11, 2012 at 2:26 PM, Lenz Emilitri lenz.lo...@gmail.comwrote:

 We are working on a project to create a general-purpose telecasting
 server - see http://wombatdialer.com - there is practically no
 documentation yet, but it's easy to set up and we tested it originating
 hundreds of channels on multiple servers. It is alpha stage, but current
 versions are free and I expect them to basically work.

 If you want to give it a shot, you can install via RPM as described on
 the website.
 Thanks
 l.


 2012/5/3 Ashish Agarwal ashisha...@gmail.com

 So what is a better approach to achieve this
 On May 3, 2012 9:20 PM, Mitul Limbani mi...@enterux.in wrote:

 The other 70 will result into failure with .call file approach.

 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel,
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-61447605
 Cell: +91-9820332422




 On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal ashisha...@gmail.comwrote:

 Hello,

 We are currently working on a project where using .call file on
 asterisk spool, outbound calls will be made from a pri line and a voice
 clip will be played.

 We know that pri has a capacity of handling only 30 channels at a
 time. Therefore, my worry is what happens if we write 100 files at a time
 on the spool. Will asterisk manage the queue or how exactly will it 
 behave.

 Regards,

 Ashish

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Re: [asterisk-users] Official numbering plan - where to get?

[Lenz Emilitri]

I had a look at the files and they are really a nightmare to parse. Some
are Word and some are Excel. Good luck :)
l.


2012/3/29 Markus unive...@truemetal.org

 http://www.itu.int/oth/T0202.aspx?parent=T0202

 But don't do it. Because I'm doing it right now. So let's not waste energy
 and do the same task twice. A complete list will soon be available, for
 free.

 And then we on this list here will start a web project to keep it updated.
 I'll let you know once the list is ready.

 I've already registered opennumberingplan.org :)


 Am 29.03.2012 11:12, schrieb Lenz Emilitri:



 DO you know if the doc files from the ITU are available somewhere for
 download?
 l.



 2012/3/22 Markus unive...@truemetal.org mailto:unive...@truemetal.org


I hope this is not too off-topic. As a kind-of follow up to rate
sheet normalization I'm still trying to figure out a solution for:
throw 10 ratesheets at a program and get the cheapest
codes/providers as output.

For this purpose I believe I need a real, detailed, accurate list of
all the dialing codes, incl. mobile codes, city codes etc. worldwide
as a reference for that particular program. There are thousands of
A-Z lists on the web, and there are thousands of codes in them, but
nothing is accurate enough or from an official source.

So, I spoke with the ITU today and they, funny enough, too don't
have such a list. At least they don't have one that is computer
parseable, like a .csv or .xls or something like that. What they
have is: a single .doc or .pdf file for EACH country (1 file per
country), which is not standardized in its content, with lots of
text and descriptions, but it has all the codes. They don't even
have such a list as a paid service. Feels like 30 years ago. :)
  Anyway, there is numberingplans.com http://numberingplans.com

which provide exactly what I'm looking for, but they don't support
one-time purchases, only subscriptions from around 100 to 990 EUR
per month, which is above my budget (and I don't need a subscription).

Does anyone have an idea where to find such a list for free, or as a
one-time purchase? If not, I'll probably go through the effort to
compile my own list based on the ITU data. Let me know in case you
want a copy then. :)

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Re: [asterisk-users] Official numbering plan - where to get?

[Lenz Emilitri]

DO you know if the doc files from the ITU are available somewhere for
download?
l.



2012/3/22 Markus unive...@truemetal.org

 I hope this is not too off-topic. As a kind-of follow up to rate sheet
 normalization I'm still trying to figure out a solution for: throw 10
 ratesheets at a program and get the cheapest codes/providers as output.

 For this purpose I believe I need a real, detailed, accurate list of all
 the dialing codes, incl. mobile codes, city codes etc. worldwide as a
 reference for that particular program. There are thousands of A-Z lists on
 the web, and there are thousands of codes in them, but nothing is accurate
 enough or from an official source.

 So, I spoke with the ITU today and they, funny enough, too don't have such
 a list. At least they don't have one that is computer parseable, like a
 .csv or .xls or something like that. What they have is: a single .doc or
 .pdf file for EACH country (1 file per country), which is not standardized
 in its content, with lots of text and descriptions, but it has all the
 codes. They don't even have such a list as a paid service. Feels like 30
 years ago. :)  Anyway, there is numberingplans.com which provide exactly
 what I'm looking for, but they don't support one-time purchases, only
 subscriptions from around 100 to 990 EUR per month, which is above my
 budget (and I don't need a subscription).

 Does anyone have an idea where to find such a list for free, or as a
 one-time purchase? If not, I'll probably go through the effort to compile
 my own list based on the ITU data. Let me know in case you want a copy
 then. :)

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Re: [asterisk-users] Collaboration Call Center Integrated with Asterisk web and email

[Lenz Emilitri]

A number of call-centers I see use the pause codes in Asterisk to mark
different types of activities, like answering to email or IM. It's not
much, but easy to implement.
l.


2012/3/27 bilal ghayyad bilmar...@yahoo.com

 Hi All;

 Is there a collaboration contact center (hope to be open source)
 Integrated with Asterisk (hope with vicidial), so the agent will be able to
 receive chat or emails sessions and deal with the customer. If the agent in
 a call with the customer, then he will not get chat session. Is there like
 this software?

 Regards
 Bilal

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Re: [asterisk-users] Asterisk 1..8 multiple queue

[Lenz Emilitri]

Shameless plug: the QueueMetrics agent page, even in the free 2-agent
version, can emulate this behavior.
You may want to check it out.
l.


2011/5/25 satish patel satish...@hotmail.com

  Hey Guys!

 We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we
 had 3 queues and we were using AgentCallbackLogin  but now its quite
 difficult to use AddQueueMember.

 Is there any easy way to logged into multiple queue using AddQueueMember ?
 and restrict agent for specific queue ?

 -S



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Re: [asterisk-users] Asterisk Queue ACD when the queues and agents has the same priority/weight

[Lenz Emilitri]

Each queue is separate and does not see what other queues are doing.
l.


2011/3/23 Marcos Setim m.se...@gmail.com

 Hello,

 I have three queues (F1,F2,F3) with default queue weight and three
 agents (A1,A2,A2) with default agent penalty. If the three agents are
 busy and tt same time a caller (C1) enter in the queue F1, and after
 20 seconds a second caller (C2) enter in the queue F2. So, few seconds
 later, the agent (A1) state comes to availabe. In this case the
 asterisk deliveries the caller (C2) to agent (A1), but the in the
 queue (F1) caller (C1) waiting time is bigger compared to caller (C2)
 of queue (F2).

 How should be the ACD behavior between queues in this case?
 How the asterisk distributes incoming calls when the queues and agents
 are the same weight/penalty?

 Thanks,


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Re: [asterisk-users] Queue pause vs logged out ?

[Lenz Emilitri]

Maybe not much from the point of view of queues, but this may make quite a
difference from the point of view of monitoring your call-center. :)
l.


2011/3/21 satish patel satish...@hotmail.com

  Hey Guys,

 I knew this is stupid question but i just want to know what is the
 difference between Queue member logged out vs Pause ?

 -Satish

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Re: [asterisk-users] How to reject an incoming call using AMI ?

[Lenz Emilitri]

Why not an unattended transfer to the queue itself, or a different queue?
l.


2011/1/10 Olivier oza_4...@yahoo.fr

 Hi,

 For a call center, I'm studying how I can offer agents the ability to
 reject an incoming call using a custom application.
 As you can guess, in this case, rejecting a call means let another agent
 answer this call (it
 doesn't mean end this call).

 The only way I could imagine for this to happen, would be to redirect the
 caller to a conference room, then hangup
 the agent call leg and then redirect the caller back to the appropriate
 queue, hoping the caller wouldn't be once again
 forwarded to the busy agent.

 Which way to implement this  would you suggest or recommend ?

 Regards



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[asterisk-users] New tutorial: Compiling Asterisk 1.8 on CentOS 64

[Lenz Emilitri]

Hello all,
as everybody else here  - I guess - I have been playing with the new
Asterisk 1.8 release. So far everything went smoothly - the compilation
phase was really straightforward, and I have a box ready for real testing
now.

I prepared a tutorial out of my experience on how to compile Asterisk 1.8
with iCal, GTalk, SNMP, MySQL, cURL and DAHDI - the usual stuff - so if
anybody is interested or has suggestions/improvements, it's here:
http://astrecipes.net/index.php?n=398 .I did not include H323 this time as I
don't have H.323 gear anymore to test it with! :)

Comments are welcome.
l.


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Re: [asterisk-users] Queue member status - BUSY

[Lenz Emilitri]

Have you tried playing with joinempty and leavewhenemèpty to avoid
people being connected to a queue with all agents in use?
l.


2010/10/20 GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com

 Hi,

 Is there a way to know if a member of a queue is currently engaged on a
 call? Or if a queue can return a busy status if all members are currently
 engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the
 scenario only falls into TIMEOUT, and has to finish the assigned number of
 seconds into the QUEUE CMD before it falls back to the next routine on the
 dialplan.

 Any ideas?

 regards,
 ryan icasiano


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Re: [asterisk-users] Solving the CDR mess of attended transfers

[Lenz Emilitri]

Is there a documentation about the CEL format?
l.


2010/9/22 Steve Murphy m...@parsetree.com


 CEL was my answer, built on the channel event goodness that Russell. It's
 now in 1.8;  but it
 lacks a converter to CDRs. You *could* just use the string of events coming
 out of CEL, but...
 I'd love to see your SQL statements to pull things together!

 I had begun writing a CEL-CDR converter, but got laid off before I could
 finish it.
 It makes a good start toward a finished package. Long ago (what, almost 2
 years now?)
 I proposed two methods of generating CDR's. One was 'simple', the other
 'Complex, or Leg Based.

 Since then, I refined the doc to just 'Simple', and outlined with some
 examples how it would/should work.
 The doc still needs to be cleaned up, but you may make sense of it.

 The trouble with CDRs is that no two shops can agree on a CDR standard that
 involves transfers, parks, etc.
 Beyond the start, answer, and end times, and some fundamental data
 about the call (source, dest,
 responsible party, etc.) There isn't much unity about what timepoints need
 to be represented, etc. And I'd seen
 so few implementations, I couldn't judge a good way to generalize the CDR
 converter.

 So, I challenge everyone to look at my simple CDR  definition, and see it
 would possible for you to adapt it
 (perhaps via a mapping from it to your desired conflagration/configuration.

 To look at the doc, do svn co
 http://svn.digium.com/svn/team/murf/asterisk-RFCs and look at the
 document in there (I have a few different formats, the .docx is the
 source).

 It's been in flux. Just the first few examples are accurate. Let me know
 what you think.

 murf



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Re: [asterisk-users] How to track a call result originated from originate AMI command

[Lenz Emilitri]

That was exactly what I was lamenting - that some common distros do not send
every event, so that AMI ends up being less than reliable. If AMi sends all
events, then it's really trivial to track calls :)
l.


2010/8/9 Motiejus Jakštys desired@gmail.com

 On Mon, Aug 9, 2010 at 12:08 PM, Lenz Emilitri lenz.lo...@gmail.com
 wrote:
  BTW, using the most common Asterisk distros out there that happen to
 sport a
  very complex dialplan, we see a lot of lost events, so that tracking
 calls
  on the basis of AMI observation alone becomes practically impossible.
  :-(
  l.

 You can filter AMI. If you know PERL, you can start with my script
 that works with callbacks:

 $callbacks{'Newstate'} = \newstate_callback;
 $callbacks{'Dial'} = \dial_callback;

 And create appropriate functions for storing desired values to the
 database. We catch Dial, Answer, Ringing, Hangup events and store that
 info to database with very accurate timestamps :-)

 http://github.com/Motiejus/Asterisk-perl-AMI/blob/master/asterisk_ami.pl

 Regards,
 Motiejus Jakštys



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Re: [asterisk-users] How to track a call result originated from originate AMI command

[Lenz Emilitri]

BTW, using the most common Asterisk distros out there that happen to sport a
very complex dialplan, we see a lot of lost events, so that tracking calls
on the basis of AMI observation alone becomes practically impossible.
:-(
l.




2010/8/8 Nasir Iqbal na...@ictinnovations.com

 Hi,

 Confusing! you are not alone here. Actually there is no unified
 development approach exist in Asterisk, every module, application introduce
 a new way to handle same things!! And the monitoring is most difficult
 part! you have to write different parsing algos to get each bit of
 information, and unfortunately you have to rewrite most of your code for
 every new release!

 And regarding your question, I recommend you to use AGI for monitoring here
 is some tips for you

- in originate command use extension as destination.
- create failed extension in same context.
- you can include some variables in originate command which can be used
later in dialplan.
- use AGI scripts in destination and failed extensions to get and
save call status in database.

 Regards



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Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

[Lenz Emilitri]

Well, actually we are in contact with quite a number of call-centers that
use the free version - a lot of times it's embedded call centers, like
internal help-desks and such. One of the nicest things of * is that you
would not buy an ACD module for a traditional pbx to support just a couple
of users, but with * it's free.
l.



2010/7/31 bruce bruce bruceb...@gmail.com

 2 users. So, it's probably never used as a free version as probably there
 are no 2 seat call centers that can survive this economy. But, it should
 great for testing.


 On Sat, Jul 31, 2010 at 10:28 AM, Leif Madsen 
 leif.mad...@asteriskdocs.org wrote:

 On 7/30/2010 5:49 AM, Lenz Emilitri wrote:
  QueueMetrics is actually free (as in beer) for very small call centers
 and
  individual hackers.

 Oh really! I didn't know that! Very nice.

 What is considered a small call centre? Are we talking up to around 5
 agents or something? Is there a limit on the number of queues?

 (I'm sure there is a page on the website that answers most of these
 questions, heh :))

 Leif Madsen.


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Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

[Lenz Emilitri]

QueueMetrics is actually free (as in beer) for very small call centers and
individual hackers.
l.

2010/7/28 Zeeshan Zakaria zisha...@gmail.com

 There is none for free.

 Zeeshan A Zakaria

 --
 www.ilovetovoip.com

 On 2010-07-27 6:12 PM, bruce bruce bruceb...@gmail.com wrote:

 :-) I knew someone would bring up FreePBX. I have FreePBX installed and
 it's not good for Queues at all. It's using the reporting tool from Areski
 and Areski has recently released an upgrade to it which again is not what I
 want.

 There are few other programs that do this but really none that are neat in
 interface or useful in features.

 I guess no one else has any thoughts on this? Maybe there is none
 available?

 Thanks,
 Bruce



 On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg dbackeb...@gmail.com
 wrote:
 
  On Mon, Jul 26...

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Re: [asterisk-users] Asterisk distribution for a Call Center

[Lenz Emilitri]

It really depends on how large your CC will be and how much money is at
stake. :-)
We have a lot of clients who are very satisfied with small call centers
based on FreePBX or Trixbox CE. Of course I would not implement a 500-seats
call center out of a standard CD.
My suggestion is: make sure you have an experienced local consultant handy
in case something goes wrong - in real life, it always does.
Just my two eurocents,
l.

2010/6/22 Alejandro Cabrera Obed aco1...@gmail.com

 Dear all, I need to build a PBX based on Asterisk for a call center. I
 have worked with raw Asterisk but it's hard to work for big
 implementations think.

 Also I have worked with Trixbox CE for a small bussines and it was
 prette good, but I have not have many features like ACD. I know there
 is another  version called Trixbox PRO -specially Call Center edition-
 that's not free but has got more features like ACD and billing.

 I've heart about AsteriskNow and I know it's free.

 What distribution/version do you recommend to me in order to implement
 a call center and taking into account I'm not an expert in programming
 from Asterisk CLI ???

 Thanks a lot

 Alejandro

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Re: [asterisk-users] Agents

[Lenz Emilitri]

Use Addmember and removemeber instead :)
l.


2010/5/14 Peter Childs pchi...@bcs.org

 I've been trying to get the hang of Agents and Queues and I must say
 its a little unclear as to how things work.

 So maybe someone has some better idea

 From what I can work out an Agent is meant to be nothing more than a
 virtual device that can be moved around physical devices... by login
 and logging out. Queues can contain any type of interface not a point
 that is partially well put in the Sark we have just got nore in the
 voi-info website It also seams to suggest that Agents are a
 deprecated feature.

 AgentLogin.

 AgentCallbackLogin is depreciated but what has it been replaced by?

 Not sure what AgentLogin is actually useful for.

 AgentCallbackLogin in the Management API does not set
 ${AGENTBYCALLERID_${CALLERID(num)}} I guessing this is a error,
 fortunatly I've worked out a way to get round it. (setvar)

 The is no way to log an agent in from the Command Line Interface.

 AgentLogoff

 Easy so long as you know the agent id you need to logoff, which means
 using ${AGENTBYCALLID_${CALLERID(num)}}

 Queues really have very little to do with Agents as any type of device
 can be statically on a queue or dynamically added when needed, but the
 info I've found seams to heavily tie the two concepts together.

 Peter

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Re: [asterisk-users] Problem with callerid(dnid) and queue

[Lenz Emilitri]

You sure it's not using the URL OPEN parameter for the very queue?
l.


2010/5/11 Carlo Dimaggio jaasmail...@gmail.com

 Hi all,

 In order to use the open url function of zoiper (it opens an url
 based on the asterisk $callerid(dnid)), I need rewriting of the dnid.
 In my dialplan I have:

 exten = 1000,3,Set(CALLERID(dnid)=newdnid)
 exten = 1000,4,Noop(${CALLERID(dnid)})
 exten = 1000,5,Queue(test-queue)

 but the callerid(dnid) shows the extension called (the member of the
 test-queue) and not the newdid. I have tried also with the option
 o in cmd Dial but without success.

 Do you know if there is a way to obtain the newdnid?


 Thanks!
 Carlo

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[asterisk-users] Check if extension loaded over AMI

[Lenz Emilitri]

Hello list,
I was wondering if there is a way to see if a given piece of dialplan is
loaded through AMI.

I have seen the GetConfig command, but it seems to expect a file name to
retrieve, and I don't necessarily know that (as it could be down the line bu
multiple levels of #includes from the main extensions.conf).

I could run an AMI Command to run the cli command dialplan show mycontext,
but I'm a bit worried by the performance cost of running a non-natively AMI
command; plus I don't love much the line-formatted response.

I could create a dummy piece of dialplan that is in the same place as . the
one i want to check, and I could try and Originate that and see if it found
or not.

All solutions above seem to be suboptimal any idea?
Thanks
l.


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Re: [asterisk-users] Evaluating Asterisk

[Lenz Emilitri]

Hello Ted,
feel free to contact us off-list - we have quite a number, from smallish to
extremely large, with varying degrees of clustering and redundancy, in
nearly any country in the world! :)
l.



2010/4/19 Ted Foote t...@abscollect.com

  I am thinking of moving from a traditional PBX to an asterisk box. Many
 of my leadership group are skeptical of asterisk. So I was hoping to find a
 call center that is currently using this technology that would not mind
 spending some time on a conference call to address some concerns that my
 team has.



 Thanks

 Ted Foote

 Allied Business Services, Inc.

 616-741-0437





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Re: [asterisk-users] Queue call to specific queuemember

[Lenz Emilitri]

Using multiple queues?
l.


2010/4/15 Asterisk Maniac asteran...@gmail.com

 Hi all,
   What would be the best way to send a call to a queue as usual, but
 telling that it should be awsered by some specific member?
 Thanks already

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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Lenz Emilitri]

Just can't wait for the live calorie counter! :)
l.


2010/4/1 Olle E. Johansson o...@edvina.net

 FOR IMMEDIATE RELEASE
 Puerto Escondido, Mexico, April 1st, 2010:

 Digium launches Asterisk VCC (TM) - a new virtual communication platform
 for enterprises, the public sector and the home.
 ===

  For VoxSwitch customers, VCCnet will mean that every user can monitor
 the movement of coworkers in realtime. By using the new APIs, additional
 data like credit card transactions, fuel consumption in the car, mileage
 in the air and calories eaten can be reported with a 3D graphical display
 using HTML5.


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Re: [asterisk-users] Asterisk system for church call center

[Lenz Emilitri]

We have a lot of clients who run small call centers based on Trixbox, and
seem to be pretty happy with them.  Have a look here:
http://queuemetrics.com/manuals/QM_Trixbox-chunked/
Thanks
l.


2010/3/31 Frank Church voi...@googlemail.com

 On 29 March 2010 21:46, Frank Church voi...@googlemail.com wrote:
  I have been asked by my church to recommend a VoIP system which can do
  the following.
 
  They do internet radio shows which are sometimes broadcast on radio.
 
  They are looking for a system which does the following for about 5
  agents, exactly as they have described it.
 
  1. Take incoming calls
 
  2. Put them on hold if there is no one to handle the call immediately,
  or transfer them to an available agent
 
  3. Take down their details, and number, (if this can be retrieved and
  saved from the caller id, thats better)
 
  4. Get them to hold on after taking their details if they still want to
 hold
 
  5. Call them back when the backlog is cleared up.
 
  I have a fairly good grasp of the hardware and programming part of
  Asterisk, having compiled it more than a few times and implemented
  A2Billing phone card and call shop system with it.
 
  But the type of software suited to the Call Center side is where my
  knowledge gap lies.
 
  I am looking for solutions based on the usual Asterisk distributions
  like AsteriskNow, trixbox, elastix etc, whether ready packaged or
  requiring additional customization.
 
 
  The matter of whether they will use soft phones, or regular phones
  with headsets is also something to consider. Soft phones with good
  GUI's may be preferred if more cost effective for them, although my
  personal preferences are with hard phones.
 
  Any recommendations - the ease of software for the end users is the
  main thing for me, and integration with the database for taking
  customers details is the main thing for me. One of the distributions
  with SugarCRM comes to mind here.
 
  Sorry for cross-posting, but ready made and commercially supported
  systems are not ruled out, if they come within their budget.
 
  Regards
 
 
  Frank Church
 

 After there response I will go with some of ready made Asterisk
 distributions, then consider to go for a commercial supported versions
 if they do not meet the churches needs.

 Thanks

 Frank




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[asterisk-users] Looping over AstDB

[Lenz Emilitri]

Hello list,
anybody has handy an example of how to loop over an ASTDB family by
getting all the keys in the dialplan?

Like I have the AstDB set as:

/test/102 : 205
/test/106 : 203
/test/113 : 209

I would like to get (in any order) the 102, 106 and 113 as members of
the family test.
TIA,
l.



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Re: [asterisk-users] Important security alert: update your dialplans now!

[Lenz Emilitri]

Yes that's cool! :)
l.


2010/2/17 Miguel Molina mmol...@millenium.com.co

 Ok, if I get it the simplest workaround would be changing this:

 exten = _X.,1,Dial(SIP/${EXTEN})

 To this:

 exten = _X.,1,Dial(SIP/${FILTER(0123456789,${EXTEN})})

 If you're intended to receive only numbers from the dialstring, right?

 See http://www.voip-info.org/wiki/view/Asterisk+func+filter

 Regards,




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Re: [asterisk-users] Important security alert: update your dialplans now!

[Lenz Emilitri]

Ok but this is available today and works fine, so it can be used as a zero
day replacement. Any syntax change is welcome but will take time until it
gets in a public release  and does not save you the hassle to change the
dialplans anyway - unless you implement it as a default behaviour at the SIP
driver level. And I got a feeling that most people will simply not bother
learning regexps
You could just as reasonably write a script to do the check, or run a check
in the dialplan itself, or change Asterisk.
l.



2010/2/15 Steve Murphy m...@parsetree.com



 On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri lenz.lo...@gmail.comwrote:

 Yes but in any case you can enter all of the strings that reasonably match
 - even if you have variable-length numbers, you will be able to determine
 that a valid number be between 5 and 15 characters - or likely 2 to 20, all
 numbers. A number of 156 characters is very likely to be a problem.


 This is probably a stupid idea, because it could only be implemented in
 trunk, and won't help with current implementations,
 and I suggested it a long time ago already when I did the fast pattern
 matching code, but I don't THINK it would be all that
 hard to offer SOME regex syntax in patterns to help reduce the impact of
 these kinds of problems.





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Re: [asterisk-users] Important security alert: update your dialplans now!

[Lenz Emilitri]

Or one could simply rewrite to:

[incoming-from-voip]
exten = XXX,1,Dial(${ext...@incoming-from-voip-old)
exten = ,1,Dial(${ext...@incoming-from-voip-old)
exten = X,1,Dial(${ext...@incoming-from-voip-old)
exten = XX,1,Dial(${ext...@incoming-from-voip-old)

[incoming-from-voip-old]
exten = _X., 1, dial(SIP/${EXTEN})

To avoid extensive rewriting and fix the current issue.
l.


2010/2/14 Steve Edwards asterisk@sedwards.com

 On Sun, 14 Feb 2010, Kyle Kienapfel wrote:

  strip_ampersands(${EXTEN})?

 (sip.conf)

 [general]
allow-characters= all
disallow-characters = 

 [example-did-provider]
allow-characters= [:numeric:]

  -



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Re: [asterisk-users] Important security alert: update your dialplans now!

[Lenz Emilitri]

Yes but in any case you can enter all of the strings that reasonably match -
even if you have variable-length numbers, you will be able to determine that
a valid number be between 5 and 15 characters - or likely 2 to 20, all
numbers. A number of 156 characters is very likely to be a problem.

BTW, you could add a catchall mail the sysadmin option - so when you get a
number that is not being matched you could be notified and adjust the
dialplan as needed.
l.



2010/2/15 Olle E. Johansson o...@edvina.net


  To avoid extensive rewriting and fix the current issue.
 That works in countries where you have fixed-length numbers. Unfortunately,
 not every dialplan works that way, so that can't be a generic advice even
 though it may solve your problems.

 Thanks for your suggestion!

 /O


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Re: [asterisk-users] Can an agent Login to a queue and be paused

[Lenz Emilitri]

In this case, I suggest you modify the login script so that your agents
always start paused. It should be trivial to do.
l.




2010/2/8 Robert Grignon rgrig...@fleetone.com

  Not a bad idea... We use queuemetrics and the login is done via Web GUI.
 I could easily just send it to pause upon login...



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Re: [asterisk-users] Can an agent Login to a queue and be paused

[Lenz Emilitri]

I'm not sure if this works for newer versions of Asterisk, but on old ones,
you could pause an agent and THEN log him on, and he'd be paused.
l.


2010/2/4 Robert Grignon rgrig...@fleetone.com


 I thought there was an option for this but cant find it

 We have a busy callcenter and I would like the agents to log in and be
 in a paused state upon login... Right now they login and they are
 instantly receiving a call

 Thanks for the input...


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