Re: [asterisk-users] Call Queues : linear strategy WITH priority

2015-08-12 Thread Jonas Kellens


On 12-08-15 16:31, A J Stiles wrote:

On Wednesday 12 Aug 2015, Jonas Kellens wrote:

Hello

I was wondering of it is possible to have Queue Agents with the same
priority (penalty) but with a certain order ?

So I have 20 Agents.

Agent 1 till Agent 10 has penalty 1.

Agent 11 till Agent 15 has penalty 2.
(only contacted if 1 -> 10 are busy)

Agent 16 till Agent 20 has penalty 3.
(only contacted if 1 -> 10 and 11 -> 15 are busy)

Within the range of Agent 1 till Agent 10, can I have a certain order in
these Agents in which they are rung ?? Like Agent 1 -> Agent 5 -> Agent
2 & 3 & 4 -> Agent 6 -> Agent 7 -> Agent 8 & 9 & 10.

What's wrong with giving agent 1 penalty 1; agent 5 penalty 2; agents 2, 3 and
4 penalty 3; agent 6 penalty 4, agent 7 penalty 5, and so forth?



By giving a different penalty to Agents 1 to 10, there is no order. With 
penalty, the Agent keeps on being contacted untill it takes the call. 
Many forget that this is how penalties work !



So in stead of going from Agent 1 to Agent 5 to Agent 2,3,4 it is very 
possible that Agent 5 keeps on ringing when Agent 1 is 'busy calling', 
in stead of going further to Agents 2,3,5.


In your case, Agent 5 will be called over and over again untill it takes 
the call.



Not exactly what I'm looking for.



Kind regards,

Jonas.

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call Queues : linear strategy WITH priority

2015-08-12 Thread A J Stiles
On Wednesday 12 Aug 2015, Jonas Kellens wrote:
> Hello
> 
> I was wondering of it is possible to have Queue Agents with the same
> priority (penalty) but with a certain order ?
> 
> So I have 20 Agents.
> 
> Agent 1 till Agent 10 has penalty 1.
> 
> Agent 11 till Agent 15 has penalty 2.
> (only contacted if 1 -> 10 are busy)
> 
> Agent 16 till Agent 20 has penalty 3.
> (only contacted if 1 -> 10 and 11 -> 15 are busy)
> 
> Within the range of Agent 1 till Agent 10, can I have a certain order in
> these Agents in which they are rung ?? Like Agent 1 -> Agent 5 -> Agent
> 2 & 3 & 4 -> Agent 6 -> Agent 7 -> Agent 8 & 9 & 10.

What's wrong with giving agent 1 penalty 1; agent 5 penalty 2; agents 2, 3 and 
4 penalty 3; agent 6 penalty 4, agent 7 penalty 5, and so forth?

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] Call Queues : linear strategy WITH priority

2015-08-12 Thread Jonas Kellens

Hello

I was wondering of it is possible to have Queue Agents with the same 
priority (penalty) but with a certain order ?


So I have 20 Agents.

Agent 1 till Agent 10 has penalty 1.

Agent 11 till Agent 15 has penalty 2.
(only contacted if 1 -> 10 are busy)

Agent 16 till Agent 20 has penalty 3.
(only contacted if 1 -> 10 and 11 -> 15 are busy)


Within the range of Agent 1 till Agent 10, can I have a certain order in 
these Agents in which they are rung ?? Like Agent 1 -> Agent 5 -> Agent 
2 & 3 & 4 -> Agent 6 -> Agent 7 -> Agent 8 & 9 & 10.




I guess I need 'linear' strategy, but will penalty option still work ?




Thank you for your feedback.

Kind regards.


Jonas.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call queues on load-balanced asterisks

2011-06-08 Thread Thomas Liu
Hi Pan & Dhaval,

In the past 8 weeks, we have delivered a load-balanced asterisks (1.4) based
call center with our flexqueue application for icson.com. It has the below
features, 

1. 2 x asterisk 1.4 boxes, 1 x mysql db box and 1 x flexqueue box(the two
are failover configured with heartbeat and custom script, and mysql
master-slave replication between two svr), 2 x kamailio boxes(failover
configured), 1 x file server boxes, 1 x app server , run freepbx &
queuemetrics. all 8 server are dell r310. 

2. the gateway is one mx100 with 4 E1 lines plugged, the incoming calls go
to kamailio2 , and routed to ast1/ast2 in round robin mode.

3. all agent phones registered to kamailio 1, and the extensions are still
maintained with freepbx

4.On asterisks, all trunks with destination to pstn or agent phones, go to
kamailio1; and incoming calls trunk from kamailio2.

5.admin also use freepbx to configure inbound routes, ivrs, announcements,
timeconditions, and recordings , etc.  the configuration files are generated
on the fly for flexqueue when apply changes. Dialplans for inbound routes
are also automatically generated and distributed to ast1 & ast2, in these
dialplan, fastagi application is installed as well to point to flexqueue.

6.flexqueue interprets the call flow configured on freepbx, and create the
queues configured on freepbx, but it's shared among all asterisk boxes. 

7.flexqueue interface with queuemetrics , and send all necessary queue logs
to queuemetrics for complete reporting & QA purpose.

8.flexqueue has a agent phone panel, and a supervisor monitoring &
management panel. Agent can logon his/her phone panel to have features like,
incoming call popup, parking, Outbound dial, hold/unhold, transfer
(cold/warm/to another queue), hangup, wrapup , pause/resume, etc. the
supervisor can logon his/her monitoring & management panel, to view realtime
event-driven agent info, queue info, and calls on-going. Besides, supervisor
can also listen to agents, barge agents' talk, and qc call records &
recordings quickly.

9.flexqueue provide web api for customer's CRM, which is asp.net based, to
make agent can click-dial in their web crm application, and playback
recordings to the agent's phone by clicking playback button beside crm
communication records. 


The above system has been put into production from today, it's fully
load-balanced asterisks based call queues or call centers. the gateways ,
the asterisk boxes can be added/removed any time. The fault asterisk box
will be detected, and bypassed from routing destinations. I wish it's a good
reference for your guys who want to create the same infrastructures.

Best Regards,

Thomas Liu

-
WShuttle Infotech Ltd.  http://www.wshuttle.com / http://www.lookmypc.com 
http://www.vicidial.cn / http://www.call-center-software.com.cn
Tel: +86 20 39230098 39230096
Mobile : +86 1390 3051 930
HK DID: +852 6950 0916, Macau DID: +853 6285 0645
Email: thomas@wshuttle.com
MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly 
Yahoo Messenger: thomaslly 
Address:  Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, 
Guangzhou Higher Education Mega Center, Guangzhou, 
Guangdong Province, China.   Zip code: 510006

--

> -Original Message-
> From: Thomas Liu [mailto:thomas@wshuttle.com]
> Sent: Wednesday, January 12, 2011 12:15 AM
> To: 'asterisk-users@lists.digium.com'
> Subject: RE: Call queues on load-balanced asterisks
> 
> Hi Pan & Dhaval,
> 
> We have implemented a FastAGI based queue with Erlang for a inbound call
> center, and call this new application as FlexQueue.
> All calls distributed on multiple asterisk boxes go through and are
controlled by
> that same remote fastagi server.
> 
> It can routing calls to any destination, by any business rules. It don't
rely on the
> db for agent/call status store & query.
> It's event driven and dict based agent/call store & query, with very good
> performance, and low cpu power consumption.
> 
> I think for your requirement, app_queue could not fulfill that.
> 
> Best Regards,
> 
> Thomas Liu
>

-
> WShuttle Infotech Ltd.  http://www.wshuttle.com /
> http://www.lookmypc.com
> http://www.vicidial.cn / http://www.call-center-software.com.cn
> Tel: +86 20 39230098 39230096
> Mobile : +86 1390 3051 930
> HK DID: +852 6950 0916, Macau DID: +853 6285 0645
> Email: thomas@wshuttle.com
> MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly
> Yahoo Messenger: thomaslly
> Address:  Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area,
> Guangzhou Higher Education Mega Center, Guangzhou,
> Guangdong Province, China.   Zip code: 510006
>

--

Re: [asterisk-users] Call queues on load-balanced asterisks

2011-01-13 Thread Pan B. Christensen

Hello Mr. Liu,

I tried searching for more information about FlexQueue, where to download 
etc. Google linked to vicidial.cn, which appears in your signature, but that 
page is all in chinese, and I couldn't find any english link. Where can I 
get more information about it? Is it a commercial product?


With kind regards,
Pan B. Christensen
Ibidium AS
http://www.ibidium.no

- Original Message - 
From: "Thomas Liu" 

To: 
Sent: Tuesday, January 11, 2011 5:14 PM
Subject: Re: [asterisk-users] Call queues on load-balanced asterisks


Hi Pan & Dhaval,

We have implemented a FastAGI based queue with Erlang for a inbound call
center, and call this new application as FlexQueue.
All calls distributed on multiple asterisk boxes go through and are
controlled by that same remote fastagi server.

It can routing calls to any destination, by any business rules. It don't
rely on the db for agent/call status store & query.
It's event driven and dict based agent/call store & query, with very good
performance, and low cpu power consumption.

I think for your requirement, app_queue could not fulfill that.

Best Regards,

Thomas Liu

-
WShuttle Infotech Ltd. http://www.wshuttle.com / http://www.lookmypc.com
http://www.vicidial.cn / http://www.call-center-software.com.cn
Tel: +86 20 39230098 39230096
Mobile : +86 1390 3051 930
HK DID: +852 6950 0916, Macau DID: +853 6285 0645
Email: thomas@wshuttle.com
MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly
Yahoo Messenger: thomaslly
Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area,
Guangzhou Higher Education Mega Center, Guangzhou,
Guangdong Province, China. Zip code: 510006

--



Hello Dhaval (and others),

As far as I can tell, realtime queue will not solve my problem. I can

statically

define the same queue with the same members on two machines as well. I was

planning

to use realtime anyway. The issue is the actual queueing of the incoming

calls.


Let?s say I define the queue "IT-support" with members Local/100 and

Local/101

on both machines. The first call comes in and is distributed by Kamailio

to Asterisk

A, and answered by 100. The next call comes in to Asterisk B, and is

answered by

101. At this point, both members are busy. Call 3 now comes in and is sent

to Asterisk

A, where it waits for a free member. Call 4 comes in and is also sent to

Asterisk

A, as is Call 5. Then call 6 is sent to Asterisk B. At this point 100

finishes

his call and becomes free. Which call is delivered to 100? As far as I can

tell,

that?s a 50/50 chance between call 3 and call 6. This is not correct

behaviour!

Call 6 should wait until calls 3, 4 and 5 (from the other server) have all

been

delivered.

In the example above: When call 3 comes in, Asterisk A may even try to

deliver

it to 101, who gets call waiting indication. He will now have two

simultaneous

calls from the same queue!

I have not found any way to share information about calls waiting in the

queue,

wait times, member states and so on between the two servers.

Unless you guys know of a way, I think I'm going to have to ask the

customer to

change their design to master-slave (with failover) instead of

load-balanced.


With kind regards,
Pan

> Hello Pan,
>
> You can user DB for this just make real time configuration of Queue and

make

> all asterisk server connected to Same DB if more load then use

replication

> for different server on DB, also So that Quque name should be same for

all

> server and asterisk can call same agent.
>
> you didnot mentioned that which purpose youwere use queue other wise i

can

> give answer in better way.
>
> regards
> Dhaval
>
> On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen 

wrote:

>
>>  Hello,
>>
>> I have been asked to implement the following design:
>>
>> Load-balanced Kamailio servers handling registrations and routing.
>> Load-balanced asterisk feature servers handling voicemail and other

things

>> Kamailio cannot do. Plus several load-balanced gateways, but they are

not

>> relevant to my question.
>>
>> All this is working fine.
>>
>> I've now been asked to start implementing calling queues, and my

question

>> is this:
>> How can I implement the same queue on multiple Asterisk servers?
>>
>> Let's say that 10 people call the same queue. These calls would then
>> currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I

make

>> Asterisk A respect the 5 people queued on the other server and vice

versa?

>>
>> Will the customer need to change their design to make the feature

servers

Re: [asterisk-users] Call queues on load-balanced asterisks

2011-01-11 Thread Thomas Liu
Hi Pan & Dhaval,

We have implemented a FastAGI based queue with Erlang for a inbound call
center, and call this new application as FlexQueue.
All calls distributed on multiple asterisk boxes go through and are
controlled by that same remote fastagi server.

It can routing calls to any destination, by any business rules. It don't
rely on the db for agent/call status store & query. 
It's event driven and dict based agent/call store & query, with very good
performance, and low cpu power consumption.

I think for your requirement, app_queue could not fulfill that. 

Best Regards,

Thomas Liu

-
WShuttle Infotech Ltd.  http://www.wshuttle.com / http://www.lookmypc.com 
http://www.vicidial.cn / http://www.call-center-software.com.cn
Tel: +86 20 39230098 39230096
Mobile : +86 1390 3051 930
HK DID: +852 6950 0916, Macau DID: +853 6285 0645
Email: thomas@wshuttle.com
MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly 
Yahoo Messenger: thomaslly 
Address:  Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, 
Guangzhou Higher Education Mega Center, Guangzhou, 
Guangdong Province, China.   Zip code: 510006

--

 
> Hello Dhaval (and others),
> 
> As far as I can tell, realtime queue will not solve my problem. I can
statically
> define the same queue with the same members on two machines as well. I was
planning
> to use realtime anyway. The issue is the actual queueing of the incoming
calls.
> 
> Let?s say I define the queue "IT-support" with members Local/100 and
Local/101
> on both machines. The first call comes in and is distributed by Kamailio
to Asterisk
> A, and answered by 100. The next call comes in to Asterisk B, and is
answered by
> 101. At this point, both members are busy. Call 3 now comes in and is sent
to Asterisk
> A, where it waits for a free member. Call 4 comes in and is also sent to
Asterisk
> A, as is Call 5. Then call 6 is sent to Asterisk B. At this point 100
finishes
> his call and becomes free. Which call is delivered to 100? As far as I can
tell,
> that?s a 50/50 chance between call 3 and call 6. This is not correct
behaviour!
> Call 6 should wait until calls 3, 4 and 5 (from the other server) have all
been
> delivered.
> 
> In the example above: When call 3 comes in, Asterisk A may even try to
deliver
> it to 101, who gets call waiting indication. He will now have two
simultaneous
> calls from the same queue!
> 
> I have not found any way to share information about calls waiting in the
queue,
> wait times, member states and so on between the two servers.
> 
> Unless you guys know of a way, I think I'm going to have to ask the
customer to
> change their design to master-slave (with failover) instead of
load-balanced.
> 
> With kind regards,
> Pan
> 
> > Hello Pan,
> >
> > You can user DB for this just make real time configuration of Queue and
make
> > all asterisk server connected to Same DB if more load then use
replication
> > for different server on DB, also So that Quque name should be same for
all
> > server and asterisk can call same agent.
> >
> > you didnot mentioned that which purpose youwere use queue other wise i
can
> > give answer in better way.
> >
> > regards
> > Dhaval
> >
> > On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen 
wrote:
> >
> >>  Hello,
> >>
> >> I have been asked to implement the following design:
> >>
> >> Load-balanced Kamailio servers handling registrations and routing.
> >> Load-balanced asterisk feature servers handling voicemail and other
things
> >> Kamailio cannot do. Plus several load-balanced gateways, but they are
not
> >> relevant to my question.
> >>
> >> All this is working fine.
> >>
> >> I've now been asked to start implementing calling queues, and my
question
> >> is this:
> >> How can I implement the same queue on multiple Asterisk servers?
> >>
> >> Let's say that 10 people call the same queue. These calls would then
> >> currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I
make
> >> Asterisk A respect the 5 people queued on the other server and vice
versa?
> >>
> >> Will the customer need to change their design to make the feature
servers
> >> master-slave with failover instead of load-balanced?
> >>
> >> Mvh
> >> Pan


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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Call queues on load-balanced asterisks

2011-01-11 Thread Danny Nicholas
  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B.
Christensen
Sent: Tuesday, January 11, 2011 5:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call queues on load-balanced asterisks

 

Hello Dhaval (and others),

 

As far as I can tell, realtime queue will not solve my problem. I can
statically define the same queue with the same members on two machines as
well. I was planning to use realtime anyway. The issue is the actual
queueing of the incoming calls.

 

Let's say I define the queue "IT-support" with members Local/100 and
Local/101 on both machines. The first call comes in and is distributed by
Kamailio to Asterisk A, and answered by 100. The next call comes in to
Asterisk B, and is answered by 101. At this point, both members are busy.
Call 3 now comes in and is sent to Asterisk A, where it waits for a free
member. Call 4 comes in and is also sent to Asterisk A, as is Call 5. Then
call 6 is sent to Asterisk B. At this point 100 finishes his call and
becomes free. Which call is delivered to 100? As far as I can tell, that's a
50/50 chance between call 3 and call 6. This is not correct behaviour! Call
6 should wait until calls 3, 4 and 5 (from the other server) have all been
delivered.

 

In the example above: When call 3 comes in, Asterisk A may even try to
deliver it to 101, who gets call waiting indication. He will now have two
simultaneous calls from the same queue!

 

I have not found any way to share information about calls waiting in the
queue, wait times, member states and so on between the two servers.

 

Unless you guys know of a way, I think I'm going to have to ask the customer
to change their design to master-slave (with failover) instead of
load-balanced.

 

With kind regards,

Pan

 

IMO your best solution to this is going to be using a database and AGI query
to keep a "quasi-real" (delayed by a few ms/sec) picture of the queue
activity.  If you kept a database on both machines and ran an AGI with each
incoming call to query queue usage on both machines or better yet, query the
queue on the remote machine and spawn a short local call to keep that agent
busy on the native machine, that would solve this issue.  Let's say that a
typical agent interaction occurs in 60 second chunks.  Call 1 comes in to
machine 1 and is answered by agent 100 as you said.  Call 2 comes into
machine 2 and is answered by 101.  When Call 3 comes in, it sees 101 and 102
as busy on both machines.  

 

You can do this, but isn't this really a Kamailio issue?

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call queues on load-balanced asterisks

2011-01-11 Thread Pan B. Christensen
Hello Dhaval (and others),

As far as I can tell, realtime queue will not solve my problem. I can 
statically define the same queue with the same members on two machines as well. 
I was planning to use realtime anyway. The issue is the actual queueing of the 
incoming calls.

Let’s say I define the queue "IT-support" with members Local/100 and Local/101 
on both machines. The first call comes in and is distributed by Kamailio to 
Asterisk A, and answered by 100. The next call comes in to Asterisk B, and is 
answered by 101. At this point, both members are busy. Call 3 now comes in and 
is sent to Asterisk A, where it waits for a free member. Call 4 comes in and is 
also sent to Asterisk A, as is Call 5. Then call 6 is sent to Asterisk B. At 
this point 100 finishes his call and becomes free. Which call is delivered to 
100? As far as I can tell, that’s a 50/50 chance between call 3 and call 6. 
This is not correct behaviour! Call 6 should wait until calls 3, 4 and 5 (from 
the other server) have all been delivered.

In the example above: When call 3 comes in, Asterisk A may even try to deliver 
it to 101, who gets call waiting indication. He will now have two simultaneous 
calls from the same queue!

I have not found any way to share information about calls waiting in the queue, 
wait times, member states and so on between the two servers.

Unless you guys know of a way, I think I'm going to have to ask the customer to 
change their design to master-slave (with failover) instead of load-balanced.

With kind regards,
Pan

> Hello Pan,
> 
> You can user DB for this just make real time configuration of Queue and make
> all asterisk server connected to Same DB if more load then use replication
> for different server on DB, also So that Quque name should be same for all
> server and asterisk can call same agent.
> 
> you didnot mentioned that which purpose youwere use queue other wise i can
> give answer in better way.
> 
> regards
> Dhaval
> 
> On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen  wrote:
>
>>  Hello,
>>
>> I have been asked to implement the following design:
>>
>> Load-balanced Kamailio servers handling registrations and routing.
>> Load-balanced asterisk feature servers handling voicemail and other things
>> Kamailio cannot do. Plus several load-balanced gateways, but they are not
>> relevant to my question.
>>
>> All this is working fine.
>>
>> I've now been asked to start implementing calling queues, and my question
>> is this:
>> How can I implement the same queue on multiple Asterisk servers?
>>
>> Let's say that 10 people call the same queue. These calls would then
>> currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make
>> Asterisk A respect the 5 people queued on the other server and vice versa?
>>
>> Will the customer need to change their design to make the feature servers
>> master-slave with failover instead of load-balanced?
>>
>> Mvh
>> Pan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call queues on load-balanced asterisks

2011-01-08 Thread DHAVAL INDRODIYA
Hello Pan,

You can user DB for this just make real time configuration of Queue and make
all asterisk server connected to Same DB if more load then use replication
for different server on DB, also So that Quque name should be same for all
server and asterisk can call same agent.

you didnot mentioned that which purpose youwere use queue other wise i can
give answer in better way.

regards
Dhaval

On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen  wrote:

>  Hello,
>
> I have been asked to implement the following design:
>
> Load-balanced Kamailio servers handling registrations and routing.
> Load-balanced asterisk feature servers handling voicemail and other things
> Kamailio cannot do. Plus several load-balanced gateways, but they are not
> relevant to my question.
>
> All this is working fine.
>
> I've now been asked to start implementing calling queues, and my question
> is this:
> How can I implement the same queue on multiple Asterisk servers?
>
> Let's say that 10 people call the same queue. These calls would then
> currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make
> Asterisk A respect the 5 people queued on the other server and vice versa?
>
> Will the customer need to change their design to make the feature servers
> master-slave with failover instead of load-balanced?
>
> Mvh
> Pan
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Call queues on load-balanced asterisks

2011-01-07 Thread Pan B. Christensen
Hello,

I have been asked to implement the following design:

Load-balanced Kamailio servers handling registrations and routing.  
Load-balanced asterisk feature servers handling voicemail and other things 
Kamailio cannot do. Plus several load-balanced gateways, but they are not 
relevant to my question.

All this is working fine.

I've now been asked to start implementing calling queues, and my question is 
this:
How can I implement the same queue on multiple Asterisk servers?

Let's say that 10 people call the same queue. These calls would then currently 
be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A 
respect the 5 people queued on the other server and vice versa?

Will the customer need to change their design to make the feature servers 
master-slave with failover instead of load-balanced?

Mvh
Pan--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call queues - issues, can't make it work.

2010-06-14 Thread Aksel Celasun
Thank You Tarek!

That was the case, and i saw now i had a typo in the extension further down, 
but, you solved it.
Now I faced a couple of other problems, alle the announcements and MOH didn’t 
play, the settings are default.
Maybe i'll figure it out.

Thank you


Regards 

Aksel


Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Tarek Sawah
Sendt: 14. juni 2010 15:00
Til: Asterisk Users
Emne: Re: [asterisk-users] Call queues - issues, can't make it work.

when you add an agent to a queue the agent should log in
try adding
member=SIP/301
member=SIP/302
instead of agent directives.
this will ring both phones.. from your output it doesn't seem to be ringing the 
agents at all.

-- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 
562 2308



From: ak...@abacus-it.no
To: asterisk-users@lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
Hello there


I have been struggling with queues, because i think this is the right module 
for our business.
My main goal, is when we receive external calls, the receptionist should be 
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.

I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5 
seconds, and then transferred to queue “teknisk”
And I thought that internal phonex/extensions 301 and 302 would ring.

But, when I ring the external number, it just rings…and rings…until it hang-ups.

CLI output shows that the commands are running, but maybe the wrong way, are 
the queue command routed to my sip provider?

Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying 
to route the queue to with the ringall argument.
This happens:
Reloading MGCP
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [4767209...@internal:1] 
NoOp("SIP/odin.service.ipallover.net-00d1", "") in new stack
-- Executing [4767209...@internal:2] 
Verbose("SIP/odin.service.ipallover.net-00d1", "Callerid num 90015103") in 
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3] 
Dial("SIP/odin.service.ipallover.net-00d1", "SIP/301,5") in new stack
  == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4] 
Queue("SIP/odin.service.ipallover.net-00d1", "teknisk") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
--  Playing 'queue-youarenext.gsm' 
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue 
position (which was 1)
--  Playing 'queue-thankyou.gsm' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
  == Spawn extension (internal, 4767209611, 4) exited non-zero on 
'SIP/odin.service.ipallover.net-00d1'

asterisk*CLI>

---
Agents.conf is default and  i have two extensions/agents
agent => 301,301
agent => 302,302


--
[r...@asterisk asterisk]# more queues.conf

[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes

member => Agent/301
member => Agent/302

-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped

;exten 301
exten => 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk);
exten => 4767209611,n,Voicemail(301);   ;Added 06.Mai.10-Aksel




Could someone please help me in the right direction here?


Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner


Re: [asterisk-users] Call queues - issues, can't make it work.

2010-06-14 Thread Tarek Sawah

when you add an agent to a queue the agent should log in try adding 
member=SIP/301member=SIP/302instead of agent directives.this will ring both 
phones.. from your output it doesn't seem to be ringing the agents at all.

-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308






From: ak...@abacus-it.no
To: asterisk-users@lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
















Hello there

 

 

I have been struggling with queues, because
i think this is the right module for our business.

My main goal, is when we receive external
calls, the receptionist should be able to transfer the call to us 

Technicians, and I am trying to add 2
extensions to a queue name [teknisk]

Extension 301 and 302.

 

I have a test setup now which I thought
should look like this:

When a external call come to my external
number (67209611) this will ring for 5 seconds, and then transferred to queue 
“teknisk”

And I thought that internal
phonex/extensions 301 and 302 would ring.

 

But, when I ring the external number, it
just rings…and rings…until it hang-ups.

 

CLI output shows that the commands are
running, but maybe the wrong way, are the queue command routed to my sip
provider?

 

Info: 67209611 is my public phone number.

90015103 is my cell phone number

301 and 302 are internal extensions in
technician department, which I am trying to route the queue to with the ringall
argument.

This happens:

Reloading MGCP

  == Using SIP RTP TOS bits 184

  == Using SIP RTP CoS mark 5

-- Executing
[4767209...@internal:1]
NoOp("SIP/odin.service.ipallover.net-00d1", "") in new
stack

-- Executing
[4767209...@internal:2]
Verbose("SIP/odin.service.ipallover.net-00d1", "Callerid num
90015103") in new stack

Callerid num 90015103

-- Executing
[4767209...@internal:3]
Dial("SIP/odin.service.ipallover.net-00d1",
"SIP/301,5") in new stack

  == Using SIP RTP TOS bits 184

 == Using SIP RTP CoS mark 5

-- Called 301

-- SIP/301-00d2 is
ringing

-- Nobody picked up in
5000 ms

-- Executing
[4767209...@internal:4]
Queue("SIP/odin.service.ipallover.net-00d1", "teknisk")
in new stack

-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'

-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1

--
 Playing 'queue-youarenext.gsm'
(language 'en')

-- Told
SIP/odin.service.ipallover.net-00d1 in teknisk their queue position (which
was 1)

--
 Playing 'queue-thankyou.gsm'
(language 'en')

-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'

-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1

  == Spawn extension (internal,
4767209611, 4) exited non-zero on 'SIP/odin.service.ipallover.net-00d1'

 

asterisk*CLI>

 

---

Agents.conf is default and  i have two
extensions/agents

agent => 301,301

agent => 302,302

 

 

--

[r...@asterisk asterisk]# more queues.conf

 

[teknisk]

music = default

announce = queue-callswaiting.gsm

strategy = ringall

timeout = 15

retry = 0

maxlen = 0

announce-frequency = 120

announce-holdtime = yes

 

member => Agent/301

member => Agent/302

 

-

Sip.conf

[301]

type=friend

secret=xx

host=dynamic

context=phones

mailbox=...@default


qualify=yes

callgroup=teknisk

-

extensions.conf snipped

 

;exten 301

exten => 4767209611,1,NoOp();

exten => 4767209611,n,Verbose(Callerid
num ${CALLERID(num)});

exten => 4767209611,n,Dial(SIP/301,5);

exten => 4767209600,n,Queue(teknisk);

exten =>
4767209611,n,Voicemail(301);  
;Added 06.Mai.10-Aksel

 

 

 

 

Could someone please help me in the right
direction here?

 

 

Med vennlig hilsen

Abacus IT AS

- din Visma Software Partner

 

Tor Aksel Celasun

Mobilnummer 900 15 103

Sentralbord/Support 4000 1850

ak...@abacus-it.no

 

  
_
The New Busy think 9 to 5 is a cute idea. Combine multiple calendars with 
Hotmail. 
http://www.windowslive.com/campaign/thenewbusy?tile=multicalendar&ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010

[asterisk-users] Call queues - issues, can't make it work.

2010-06-14 Thread Aksel Celasun
Hello there


I have been struggling with queues, because i think this is the right module 
for our business.
My main goal, is when we receive external calls, the receptionist should be 
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.

I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5 
seconds, and then transferred to queue "teknisk"
And I thought that internal phonex/extensions 301 and 302 would ring.

But, when I ring the external number, it just rings...and rings...until it 
hang-ups.

CLI output shows that the commands are running, but maybe the wrong way, are 
the queue command routed to my sip provider?

Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying 
to route the queue to with the ringall argument.
This happens:
Reloading MGCP
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [4767209...@internal:1] 
NoOp("SIP/odin.service.ipallover.net-00d1", "") in new stack
-- Executing [4767209...@internal:2] 
Verbose("SIP/odin.service.ipallover.net-00d1", "Callerid num 90015103") in 
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3] 
Dial("SIP/odin.service.ipallover.net-00d1", "SIP/301,5") in new stack
  == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4] 
Queue("SIP/odin.service.ipallover.net-00d1", "teknisk") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
--  Playing 'queue-youarenext.gsm' 
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue 
position (which was 1)
--  Playing 'queue-thankyou.gsm' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
  == Spawn extension (internal, 4767209611, 4) exited non-zero on 
'SIP/odin.service.ipallover.net-00d1'

asterisk*CLI>

---
Agents.conf is default and  i have two extensions/agents
agent => 301,301
agent => 302,302


--
[r...@asterisk asterisk]# more queues.conf

[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes

member => Agent/301
member => Agent/302

-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped

;exten 301
exten => 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk);
exten => 4767209611,n,Voicemail(301);   ;Added 06.Mai.10-Aksel




Could someone please help me in the right direction here?


Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner

Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call queues

2007-07-10 Thread Floyd
Floyd wrote:
> Hi everyone:
> 
> I've searching for a while and haven't found what i
> need.
> The thing is that i have a tdm422p with the two fxo
> ports connected to the pstn. I want my sip users to
be
> able to call other numbers(any number) in the pstn
> through my zap fxo channels. I have a big number of
> sip users so as you can imagine there will be
> congestion when some of them(more than two!!) want
to
> call outside, that is why i want to be able to put
> those outgoing calls in a queue. For example if i
want
> to call someone in the pstn and the fxo port is
> already in use, i want to be placed in a queue and
> when the channel is free my call is routed to the
> aproppiated destination. As far as i have read the
> queues are not for this kind of stuffs,  there are
> just agents or extensions that attend the calls in
the
> queue and nothing more. am i wrong???
> Any help will be useful. 
> thanks in advance!!


Hi,
first of all i would like to thanks to C. Chad Wallace
Noah  and Rob Schall. I just solve the problem of the
outgoing call queue. Following the instructions from
Chad 
i did something like this in my extensions.conf:

exten => _9XXX,1,Answer
exten => _9XXX,2,Set(_number=${EXTEN:1})
exten => _9XXX,3,Wait(2)
exten => _9XXX,4,NoOP(${number})
exten => _9XXX,5,Queue(Myqueue)
exten => _9XXX,6,Hangup

I also have a context like this

[outbound]

exten => 1,1,NoOP(${number})
exten => 1,2,Dial(Zap/G2/${number},30,t)

exten => 2,1,NoOP(${number})
exten => 2,2,Dial(Zap/G2/${number},30,t)


Finally in queues.conf

 [myqueue]
member => Local/[EMAIL PROTECTED]/n
member => Local/[EMAIL PROTECTED]/n

And it worked perfect!!

I have my outgoing calls routed ok and the variable
travells throgh the queue without problems..


thanks!!

Eve



__
Correo Yahoo!
Espacio para todos tus mensajes, antivirus y antispam ¡gratis! 
Regístrate ya - http://correo.espanol.yahoo.com/ 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Call Queues

2007-07-05 Thread C. Chad Wallace
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Floyd wrote:
> Hi everyone:
> 
> I've searching for a while and haven't found what i
> need.
> The thing is that i have a tdm422p with the two fxo
> ports connected to the pstn. I want my sip users to be
> able to call other numbers(any number) in the pstn
> through my zap fxo channels. I have a big number of
> sip users so as you can imagine there will be
> congestion when some of them(more than two!!) want to
> call outside, that is why i want to be able to put
> those outgoing calls in a queue. For example if i want
> to call someone in the pstn and the fxo port is
> already in use, i want to be placed in a queue and
> when the channel is free my call is routed to the
> aproppiated destination. As far as i have read the
> queues are not for this kind of stuffs,  there are
> just agents or extensions that attend the calls in the
> queue and nothing more. am i wrong???
> Any help will be useful. 
> thanks in advance!!

You could probably do this using the Local channel.  You'd create a
context, say outbound, to take calls from the queue and connect them to
a Zap channel, with 2 extensions in that context--one for each channel.
 Then you add each of those extensions as members of the queue:

member => Local/[EMAIL PROTECTED]/n
member => Local/[EMAIL PROTECTED]/n

Make sure your dialplan in outbound returns Busy if the Zap channel is busy.

The tricky part would be passing the dialed number through...  But if
you set an inheriting channel var, it should go through the queue and
into the Local channel to your outbound extension.

Sorry I don't have any code for you... I haven't done it yet; I'm just
putting the idea out there.

Hope this helps!
Good luck.

- --

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0


-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFGjWCTKeSNHCYiCKARAtdgAKCVUs6OF2KIpjbpwQFrwr2E4NatVACfWh6I
9XwYqQ7cc5gwVznybIglBGs=
=miEL
-END PGP SIGNATURE-

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Call Queues

2007-07-05 Thread Rob Schall

Noah Miller wrote:

Hi Eve -

  

The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
congestion when some of them(more than two!!) want to
call outside, that is why i want to be able to put
those outgoing calls in a queue. For example if i want
to call someone in the pstn and the fxo port is
already in use, i want to be placed in a queue and
when the channel is free my call is routed to the
aproppiated destination. As far as i have read the
queues are not for this kind of stuffs,  there are
just agents or extensions that attend the calls in the
queue and nothing more. am i wrong???



I think your suspicions may be correct.  You could add your ZAP
channels as members in queues.conf, maybe something like this: members
=> ZAP/1, and then use queue() on your outbound extensions.  The
problem is how will your agents, in this case your ZAP trunks, know to
"pick up the line" when they are not busy.  You'd have to get these
lines to somehow go offhook if they're not already busy.  Maybe you
can do this with an AGI script.  I don't know, I've never tried to
artificially control hook status.

Personally, I'd probably just skip the whole queue idea and get some
cheap SIP or IAX trunks and fall back to them when the ZAP lines are
busy.


- Noah

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  
Possibly do a combination of things. Check if those zap chans are in 
use/busy. If they care, then create a call file using a script. I 
haven't played too much with it, so I don't know if those will queue 
until they can complete or if it will just error and delete itself. If 
you really are determined, you might even be able to route all requests 
to a script. Then have it check if there are any open lines... if so, 
create the call file... if not, then put it in a queue (in python, 
etc... not an asterisk queue), and try again in a min and see if a 
channel has opened up.


Disclaimer - I have no idea if this idea will work. :)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call Queues

2007-07-05 Thread Noah Miller
Hi Eve -

> The thing is that i have a tdm422p with the two fxo
> ports connected to the pstn. I want my sip users to be
> able to call other numbers(any number) in the pstn
> through my zap fxo channels. I have a big number of
> sip users so as you can imagine there will be
> congestion when some of them(more than two!!) want to
> call outside, that is why i want to be able to put
> those outgoing calls in a queue. For example if i want
> to call someone in the pstn and the fxo port is
> already in use, i want to be placed in a queue and
> when the channel is free my call is routed to the
> aproppiated destination. As far as i have read the
> queues are not for this kind of stuffs,  there are
> just agents or extensions that attend the calls in the
> queue and nothing more. am i wrong???

I think your suspicions may be correct.  You could add your ZAP
channels as members in queues.conf, maybe something like this: members
=> ZAP/1, and then use queue() on your outbound extensions.  The
problem is how will your agents, in this case your ZAP trunks, know to
"pick up the line" when they are not busy.  You'd have to get these
lines to somehow go offhook if they're not already busy.  Maybe you
can do this with an AGI script.  I don't know, I've never tried to
artificially control hook status.

Personally, I'd probably just skip the whole queue idea and get some
cheap SIP or IAX trunks and fall back to them when the ZAP lines are
busy.


- Noah

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] Call Queues

2007-07-05 Thread Floyd
Hi everyone:

I've searching for a while and haven't found what i
need.
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
congestion when some of them(more than two!!) want to
call outside, that is why i want to be able to put
those outgoing calls in a queue. For example if i want
to call someone in the pstn and the fxo port is
already in use, i want to be placed in a queue and
when the channel is free my call is routed to the
aproppiated destination. As far as i have read the
queues are not for this kind of stuffs,  there are
just agents or extensions that attend the calls in the
queue and nothing more. am i wrong???
Any help will be useful. 
thanks in advance!!

eve

__
Correo Yahoo!
Espacio para todos tus mensajes, antivirus y antispam ¡gratis! 
Regístrate ya - http://correo.espanol.yahoo.com/ 

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call queues, agents with DND status set.

2005-12-04 Thread lenz
there should be a way in agents.conf to autologoff agents after a while  
the do not answer the phone.

l.


In data Sat, 03 Dec 2005 23:48:05 +0100, Vladimir S. Blazhkun  
<[EMAIL PROTECTED]> ha scritto:




-- Called 1101
-- Agent/1101 is ringing
-- Got SIP response 480 "Temporarily Unavailable" back from x.x.x.x
-- SIP/1101-9b08 is circuit-busy

Is it possible to force logoff such agents?





--
Assum est, versa et manduca.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call queues, agents with DND status set.

2005-12-03 Thread Vladimir S. Blazhkun


-- Called 1101
-- Agent/1101 is ringing
-- Got SIP response 480 "Temporarily Unavailable" back from x.x.x.x
-- SIP/1101-9b08 is circuit-busy

Is it possible to force logoff such agents?

--
Vladimir S. Blazhkun,   Personal Communications Systems, LLC.
Leading IP NCC Specialist,  Work phone: +7 095 7847617.
JNCIA-M #773, JNCIS-M #1100.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call Queues

2005-08-25 Thread Elmar Haneke

Hi,

I do have two questions regarding call queues:

1) How can I reach that waiting calls are also removed on removing the 
last agent listening to the queue. All I found is the switch to 
prevent new calls enter the queue after the last agent left.


2) Currently my queue does ring the agent after playing the "you are 
first". How can I have the phone start ringing while the message is 
played?


Elmar
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call Queues and Agent Call Logs/Wrapup logs

2005-08-13 Thread Tom Rymes
Now that we have a well functioning Asterisk system that queues our
calls and distributes them to our CSRs, I would like to implement a
better system for our agents to keep a log of all of their calls, which
we currently do using MS Word. (As you would expect, this is a less than
ideal solution!)

I am looking for a simple program that will allow our agents to enter
notes on each call that they take and save it to a database. Basically,
The call would come in, and I would use something like astGUIClient or
IPSwitchboard to perform a screenpop to a web page or program that would
let the agent type in notes re: their call. Something along the lines
of: "Sally Jones called re: Account #1234 and wanted to know what her
balance was and if she could make a payment by credit card. I took her
information, charged the card and faxed her the receipt."

Ideally, the note would be saved with the CDR data so I could search by
extension, date, etc and find all relevant entries.

Earlier in the month something similar was mentioned, and the poster
mentioned searching Google, but I have yet to find an appropriate
solution, and before I go and try to reinvent this wheel, I thought I
would ask the list members if they have implemented something similar.
Please let me know if you have any suggestions.

Thank you for your consideration,

Tom



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call queues bug?

2005-08-11 Thread Neil Bullock
Please has anyone experienced a bug with queues in Asterisk?

No matter what settings my queue always thinks it got agents available
in it. Plus if I take all the members out, then calls don't join the
queue even though I've specified join-empty.

Any advice?

Neil

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call queues

2005-08-10 Thread Shaun Dwyer

Hi,

I'm want to do something slighty different with call queues than the 
config allows...


I wish to have things work in an 'overflowish' manner. Ie, it works just 
like 'roundrobin', where it rings
on one phone, no answer, rings on the next etc etc, except I want it to 
keep ringing on all phones that have
rung so far. For example, call comes in, phone 1 rings, in 15 seconds, 
phone 2 starts to ring while phone 1 keeps
rining. in a further 15 seconds, phone 3 starts to ring while 1 and 2 
are still ringing etc etc.


I'd code it in myself, but I'm not a C coder :)

Does anyone have any solutions for this? The only way I personally could 
potentially do something like this is with

some dialplan magic and an AGI script perhaps.

Cheers,
-Shaun
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] call queues problem

2005-06-04 Thread Guido Hecken
I think you have missed something with your agents.conf and with the member
lines in queues.conf.

This works for us:

In queues.conf:

[gws-wartefeld]

music = default
strategy = ringall
context = queue-out
timeout = 15
wrapuptime=10
announce-frequency = 0
announce-holdtime = no
queue-youarenext = queue-youarenext ;   ("You are now first
in line.")
queue-thereare  = queue-thereare;   ("There are")
queue-callswaiting = queue-callswaiting ;   ("calls waiting.")
queue-holdtime = queue-holdtime ;   ("The current est. holdtime
is")
queue-minutes = queue-minutes   ;   ("minutes.")
queue-seconds = queue-seconds   ;   ("seconds.")
queue-thankyou = queue-thankyou ;   ("Thank you for your
patience.")
queue-lessthan = queue-less-than;   ("less than")
queue-reporthold = queue-reporthold ;   ("Hold time")
joinempty = no
member => Agent/6301
member => Agent/6302
member => Agent/6303
member => Agent/6304
member => Agent/6305

In agents.conf:

[general]

persistentagents=yes
[agents]
ackcall=no
musiconhold => default
updatecdr=yes
agent => 6301,,Agent 1
agent => 6302,,Agent 2
agent => 6303,,Agent 3
agent => 6304,,Agent 4
agent => 6305,,Agent 5

Hope this helps

Regards

Guido Hecken

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] call queues problem

2005-06-04 Thread Sander








Hi there i just setup a asterisk box with
autoattendant and call queues, but it seems that when one of the agents is busy
all the new calls will stay on hold until

The agent hangs up then all phone will ring  

 

 

 

[aftersales]

 

musiconhold = default

timeout = 15

retry = 5

maxlen = 0

 

member => sip/131

member => sip/132

member => sip/133

member => sip/134

 

 

this is the queue

 

hope someone can help me here 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Call Queues and Transfers

2005-03-15 Thread Anton Krall
On dial command yes, wtWT just in case, and it works when Im the one that
originated the call, but for example, I have the same problem that you have
when the call comes in thru a Zap channel. I cant make transfer to work
eventhough the dial command that sent the incoming Zap call to me has wtWT. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guido Hecken
Sent: Martes, 15 de Marzo de 2005 04:22 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Queues and Transfers

We had the same problems with transferring calls in queues.
Sometimes, after pressing the # Key twice !!, we hear Allison say
"Transferring".
Which Phones do you use?
What shows up in the cli debug?
Are you using t and T options in the dial command?

Regards,

Guido Hecken

> Guys.. Why is it that when a call comes to a call queue and in term 
> gets assigned to an agent, if that agent tries to xfer the call using 
> # or any other feature, it doesn't do anything? I just hear the 
> "pleeps" on the
phone
> but asterisk doesn't intervene with the "Transfer" prompt.
> 
> Am I missing something?
> 
> Thx!

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Call Queues and Transfers

2005-03-15 Thread Anton Krall



Jolio, no, I checked the wiki and didnt see that parameter 
there, but I just checked show application queue and made the necessary 
modifications. 
 
Thx Guys!


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of João 
AmaroSent: Martes, 15 de Marzo de 2005 04:12 a.m.To: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Call Queues and Transfers
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1HiHow 
do you queue the incoming call ?Do you queue the call with the  t 
option (allow the called user totransfer the calling user) 
?RegardsJoão AmaroAnton Krall wrote:| Guys.. Why 
is it that when a call comes to a call queue and in term| gets assigned to 
an agent, if that agent tries to xfer the call| using # or any other 
feature, it doesn't do anything? I just hear| the "pleeps" on the phone but 
asterisk doesn't intervene with the| "Transfer" prompt.|| Am I 
missing something?|| Thx!||-BEGIN PGP 
SIGNATURE-Version: GnuPG v1.2.4 
(GNU/Linux)iD8DBQFCNrUHJUm/Bor63CERArJ7AJ9abH3agaqqq12Gc4HIl+Y5wlVY/wCeM/PO7WSe4JfZVshJVbAqPC/4r40==EajH-END 
PGP SIGNATURE-
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Call Queues and Transfers

2005-03-15 Thread Anton Krall
Thx Adam, Ill try with |t and see what happens.

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev
Sent: Martes, 15 de Marzo de 2005 04:08 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Queues and Transfers

On Tue, 2005-03-15 at 03:53 -0600, Anton Krall wrote:
> Guys.. Why is it that when a call comes to a call queue and in term 
> gets assigned to an agent, if that agent tries to xfer the call using 
> # or any other feature, it doesn't do anything? I just hear the 
> "pleeps" on the phone but asterisk doesn't intervene with the "Transfer"
prompt.
> 
> Am I missing something?

Yes, all the information that would allow me to actually help you...

How about the output from the * CLI before/during/after this attempted
transfer. How about a copy of the extensions.conf (relevant portions)?
That's just for starters... 

PS, oh okay, I'll give you a fish "show application queue"

Regards,
Adam

--
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Call Queues and Transfers

2005-03-15 Thread Guido Hecken
We had the same problems with transferring calls in queues.
Sometimes, after pressing the # Key twice !!, we hear Allison say
"Transferring".
Which Phones do you use?
What shows up in the cli debug?
Are you using t and T options in the dial command?

Regards,

Guido Hecken

> Guys.. Why is it that when a call comes to a call queue and in term gets
> assigned to an agent, if that agent tries to xfer the call using # or any
> other feature, it doesn't do anything? I just hear the "pleeps" on the
phone
> but asterisk doesn't intervene with the "Transfer" prompt.
> 
> Am I missing something?
> 
> Thx!

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call Queues and Transfers

2005-03-15 Thread João Amaro




-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi

How do you queue the incoming call ?
Do you queue the call with the  t option (allow the called user to
transfer the calling user) ?

Regards

João Amaro

Anton Krall wrote:

| Guys.. Why is it that when a call comes to a call queue and in term
| gets assigned to an agent, if that agent tries to xfer the call
| using # or any other feature, it doesn't do anything? I just hear
| the "pleeps" on the phone but asterisk doesn't intervene with the
| "Transfer" prompt.
|
| Am I missing something?
|
| Thx!
|
|
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)

iD8DBQFCNrUHJUm/Bor63CERArJ7AJ9abH3agaqqq12Gc4HIl+Y5wlVY/wCeM/PO
7WSe4JfZVshJVbAqPC/4r40=
=EajH
-END PGP SIGNATURE-



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Call Queues and Transfers

2005-03-15 Thread Adam Goryachev
On Tue, 2005-03-15 at 03:53 -0600, Anton Krall wrote:
> Guys.. Why is it that when a call comes to a call queue and in term gets
> assigned to an agent, if that agent tries to xfer the call using # or any
> other feature, it doesn't do anything? I just hear the "pleeps" on the phone
> but asterisk doesn't intervene with the "Transfer" prompt.
> 
> Am I missing something?

Yes, all the information that would allow me to actually help you...

How about the output from the * CLI before/during/after this attempted
transfer. How about a copy of the extensions.conf (relevant portions)?
That's just for starters... 

PS, oh okay, I'll give you a fish "show application queue"

Regards,
Adam

-- 
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call Queues and Transfers

2005-03-15 Thread Anton Krall
Guys.. Why is it that when a call comes to a call queue and in term gets
assigned to an agent, if that agent tries to xfer the call using # or any
other feature, it doesn't do anything? I just hear the "pleeps" on the phone
but asterisk doesn't intervene with the "Transfer" prompt.

Am I missing something?

Thx!

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call Queues, CallerID, SIP and AutoDial

2004-09-11 Thread R Wong



Hello,
 
Current moment, 
I've successfully put the incoming calls into Queues and dial to 
an idle agents. When the agents answer the calls,  the 
agents can hear the pre-recorded message to incidate what's the service that the 
call is calling.
But there one problem that I'm not able  to 
make it having the Caller ID display on the X-Lite. Even I try to make a call 
direct transfer from Asterisk to SIP X-Lite, it's not displaying the CallerID 
too!
Is there's any method I can show the CallerID on 
the X-Lite?
 
By the way, I also want to know how can I make the 
Auto Dial depends on current idle agents? I know that when I put the .call files 
into /var/spool/asterisk/outgoing, the Asterisk will make outbound call 
regardness how many files there...any suggestion?
 
Thanks in advance!
 
 
Regards,
 
R Wong
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Call queues

2004-07-23 Thread avizion
hehe - well - I think you missed the word "persistant" :)

After a few hours of digging in old docs - I also found the "new" 3rd parameter
in a more recent doc. Not sure how I missed in the first place - but I did.

Never the less it did not solve my problem of having members added to my queues
without using AgentLogin but still have a penalty.

While goofing around I found out that adding an actual client to the queues.conf
would give me a startup penalty (i.e. member => IAX2/513,50). However - this
penalty is lost when the use call app RemoveQueueMember and never set again
when they call AddQueueMember.

So I basically just wrote a small perl script to tape through queues.conf
looking up the client and add the penalty to the AddQueueMember call from
extensions.conf.

What I would like to have though - was the app_queue to do this for me. The code
is almost there I'm sure. It just needs to lookup the penalty for a dynamically
added user - if defined in queues.conf.

PS: I didn't even make my wrapper look for Agent/Ext or IAX2/Ext, just "Ext" -
but is works fine for my use atm. Only thing missing is multiple queue support
- but that's easy.

Are you still laughing? Please tell me better ways... I seached high and low for
this - and didn't find anything. And seeing time is an issue here - I didn't
want to hack the actual app_queue.c yet.

But as I mentioned - I will most likely take this to -dev list and
bugs.digium.com when ready.

Thank you for your time and comment :)

--
avizion on irc.freenode.org #asterisk

Quoting Troy Settle <[EMAIL PROTECTED]>:
> Avizion,  you're joking right?
>AddQueueMember(queuename[|interface[|penalty]]):
> The AddQueueMember function does indeed allow you to set the penalty.
>
> Too bad penalties don't work though (or maybe they work too well?)
>
> SIP/100, penalty 1
> SIP/200, penalty 2
>
> Call comes in, SIP/100 picks up
>
> Call comes in, SIP/100 is busy, but SIP/200 NEVER rings...

> >
> > Quoting Jeremy Kenney <[EMAIL PROTECTED]>:
> > > Hello I am new to asterisk I want to setup the call queues
> > where it will
> > > ring multiple devices at the same time and send the call to
> > the first one
> > > that is picked up.  There doesn't need to be an agent login
> > for this I don't
> > > think I just want setup so no login is required.  Please help
> >
> > There are several ways to accomplish this.
> >
> > Like the two others posts suggest - you can simply use the
> > Dial() application
> > directly. This will leave you with exactly the functionality
> > you are asking
> > for. What is does not give you is a real queue where members
> > can join / part as
> > they see fit (app. AddQueueMember / RemoveQueueMember). If
> > you want to have
> > your "agents" logged in from the start, you can simply define these in
> > etc/queues.conf like "SIP/phone1" or "IAX2/phone1". The last
> > option will even
> > let you define a penalty (in etc/queues.conf).
> >
> > What this lacks is a persistant penalty. I've been using a little time
> > investigating this - and I came to the conclusion that if I
> > want persistant
> > penalties for dynamically added members I would have to write
> > my own wrapper in
> > AGI. While I'm pretty much done with that part - it's not
> > exactly a beautiful
> > hack - but I might publish it if wanted.
> >
> > I will be posting on the asterisk-dev list soon - in order to
> > get second
> > oppinions on this implementation. Several things needs
> > coverage - but all this
> > in due time :)
> >
> > I hope you can use this - and feel free to ask into any of
> > the above...--
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Call queues

2004-07-23 Thread Troy Settle

Avizion,  you're joking right?

  -= Info about application 'AddQueueMember' =- 

[Synopsis]:
Dynamically adds queue members

[Description]:
   AddQueueMember(queuename[|interface[|penalty]]):

The AddQueueMember function does indeed allow you to set the penalty.

Too bad penalties don't work though (or maybe they work too well?)

SIP/100, penalty 1
SIP/200, penalty 2

Call comes in, SIP/100 picks up

Call comes in, SIP/100 is busy, but SIP/200 NEVER rings...

*sigh*

--
  Troy Settle
  Pulaski Networks
  http://www.psknet.com
  866.477.5638
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of avizion
> Sent: Friday, July 23, 2004 5:54 AM
> To: [EMAIL PROTECTED]; Jeremy Kenney
> Cc: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Call queues
> 
> Quoting Jeremy Kenney <[EMAIL PROTECTED]>:
> > Hello I am new to asterisk I want to setup the call queues 
> where it will
> > ring multiple devices at the same time and send the call to 
> the first one
> > that is picked up.  There doesn't need to be an agent login 
> for this I don't
> > think I just want setup so no login is required.  Please help
> 
> There are several ways to accomplish this.
> 
> Like the two others posts suggest - you can simply use the 
> Dial() application
> directly. This will leave you with exactly the functionality 
> you are asking
> for. What is does not give you is a real queue where members 
> can join / part as
> they see fit (app. AddQueueMember / RemoveQueueMember). If 
> you want to have
> your "agents" logged in from the start, you can simply define these in
> etc/queues.conf like "SIP/phone1" or "IAX2/phone1". The last 
> option will even
> let you define a penalty (in etc/queues.conf).
> 
> What this lacks is a persistant penalty. I've been using a little time
> investigating this - and I came to the conclusion that if I 
> want persistant
> penalties for dynamically added members I would have to write 
> my own wrapper in
> AGI. While I'm pretty much done with that part - it's not 
> exactly a beautiful
> hack - but I might publish it if wanted.
> 
> I will be posting on the asterisk-dev list soon - in order to 
> get second
> oppinions on this implementation. Several things needs 
> coverage - but all this
> in due time :)
> 
> I hope you can use this - and feel free to ask into any of 
> the above...
> 
> Regards
> 
> - avizion on irc.freenode.org #asterisk
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call queues

2004-07-23 Thread avizion
Quoting Jeremy Kenney <[EMAIL PROTECTED]>:
> Hello I am new to asterisk I want to setup the call queues where it will
> ring multiple devices at the same time and send the call to the first one
> that is picked up.  There doesn't need to be an agent login for this I don't
> think I just want setup so no login is required.  Please help

There are several ways to accomplish this.

Like the two others posts suggest - you can simply use the Dial() application
directly. This will leave you with exactly the functionality you are asking
for. What is does not give you is a real queue where members can join / part as
they see fit (app. AddQueueMember / RemoveQueueMember). If you want to have
your "agents" logged in from the start, you can simply define these in
etc/queues.conf like "SIP/phone1" or "IAX2/phone1". The last option will even
let you define a penalty (in etc/queues.conf).

What this lacks is a persistant penalty. I've been using a little time
investigating this - and I came to the conclusion that if I want persistant
penalties for dynamically added members I would have to write my own wrapper in
AGI. While I'm pretty much done with that part - it's not exactly a beautiful
hack - but I might publish it if wanted.

I will be posting on the asterisk-dev list soon - in order to get second
oppinions on this implementation. Several things needs coverage - but all this
in due time :)

I hope you can use this - and feel free to ask into any of the above...

Regards

- avizion on irc.freenode.org #asterisk
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call queues

2004-07-23 Thread Jason Williams
On Fri, 23 Jul 2004 02:26:26 -0400, Jeremy Kenney <[EMAIL PROTECTED]> wrote:
> Hello I am new to asterisk I want to setup the call queues where it will
> ring multiple devices at the same time and send the call to the first one
> that is picked up.  There doesn't need to be an agent login for this I don't
> think I just want setup so no login is required.  Please help


Check out the dial command 

Show application dial

dial(device1&device2&device3)


Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call queues

2004-07-22 Thread Jeremy Kenney
Hello I am new to asterisk I want to setup the call queues where it will
ring multiple devices at the same time and send the call to the first one
that is picked up.  There doesn't need to be an agent login for this I don't
think I just want setup so no login is required.  Please help

-Jeremy

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Call Queues help

2004-07-15 Thread brian
queue-youarenext = queue-youarenext

Like that.. remove the quotes and all the crap at the end.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kyle Hagan
> Sent: Thursday, July 15, 2004 4:48 PM
> To: Asterisk
> Subject: [Asterisk-Users] Call Queues help
>
> I've got the call cuing all setup and working, but im trying to get the
> Callswaiting,you are caller #, etc, and its not working.
>
> I have the following inthere as stated:
>
> queue-youarenext = "queue-youarenext" ("You are now first in line.")
> queue-thereare = "queue-thereare" ("There are")
> queue-callswaiting = "queue-callswaiting" ("calls waiting.")
> queue-holdtime = "queue-holdtime" ("The current est. holdtime is")
> queue-minutes = "queue-minutes" ("minutes.")
> queue-thankyou = "queue-thankyou" ("Thank you for your patience.")
>
> but i get:
>
> Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File
> "queue-thereare" ("There are") does not exist in any format
> Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable
> to open "queue-thereare" ("There are") (format GSM): No such file or
> directory
> -- Playing 'digits/2' (language 'en')
> Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File
> "queue-callswaiting" ("calls waiting.") does not exist in any format
> Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable
> to open "queue-callswaiting" ("calls waiting.") (format GSM): No such
> file or directory
> -- Hold time for queue1 is 0 minutes
> -- Told Local/[EMAIL PROTECTED],2 in queue1 their queue position (which
> was 2)
> Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File
> "queue-thankyou" ("Thank you for your patience.") does not exist in any
> format
> Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable
> to open "queue-thankyou" ("Thank you for your patience.") (format GSM):
> No such file or directory
>
> I've downloaded the asterisk additional sound files and installed them
> and tried to use those instead and I can play them with the playback
> command, but they dont work here either.
>
> Kyle
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call Queues help

2004-07-15 Thread Kyle Hagan
I've got the call cuing all setup and working, but im trying to get the 
Callswaiting,you are caller #, etc, and its not working.

I have the following inthere as stated:
queue-youarenext = "queue-youarenext" ("You are now first in line.") 
queue-thereare = "queue-thereare" ("There are") 
queue-callswaiting = "queue-callswaiting" ("calls waiting.") 
queue-holdtime = "queue-holdtime" ("The current est. holdtime is") 
queue-minutes = "queue-minutes" ("minutes.") 
queue-thankyou = "queue-thankyou" ("Thank you for your patience.")

but i get:
Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File 
"queue-thereare" ("There are") does not exist in any format
Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable 
to open "queue-thereare" ("There are") (format GSM): No such file or 
directory
   -- Playing 'digits/2' (language 'en')
Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File 
"queue-callswaiting" ("calls waiting.") does not exist in any format
Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable 
to open "queue-callswaiting" ("calls waiting.") (format GSM): No such 
file or directory
   -- Hold time for queue1 is 0 minutes
   -- Told Local/[EMAIL PROTECTED],2 in queue1 their queue position (which 
was 2)
Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File 
"queue-thankyou" ("Thank you for your patience.") does not exist in any 
format
Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable 
to open "queue-thankyou" ("Thank you for your patience.") (format GSM): 
No such file or directory

I've downloaded the asterisk additional sound files and installed them 
and tried to use those instead and I can play them with the playback 
command, but they dont work here either.

Kyle
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call Queues, Call groups

2004-04-23 Thread Paul Mahler
Is anyone successfully using call queues and call groups? If so do you have
an example configuration? 
 
The wicki and mailing list information I found is pretty old. 
 
Thanks!
 
Paul
[EMAIL PROTECTED]
 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call Queues

2004-02-14 Thread Jon Stockill
On Tue, 10 Feb 2004, Jonathan Stanton @ Home wrote:

> Any ideas / sugestions welcome.

Having the queue calls delivered to an agent login would appear to be the
easiest way to do it - just log in the agent on any phone you like, and
calls will be diverted to that phone. As another poster noted, one
little gotcha is that if your announcement file is missing, the calls
will be disconnected when you try to answer them.

-- 
Jon Stockill
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Call Queues

2004-02-12 Thread B. J. Bomar
Title: Message



Here 
is some config that I cooked up.  It may be a little rough around the 
edges, and it incorporates multiple users.
 
exten 
=> *801,1,Answerexten => 
*801,2,SetVar(temp=${loggedin${CALLERIDNUM}})exten => 
*801,3,GotoIf($[${temp} = 1]?50:)exten => *801,4,GotoIf($[${CALLERIDNUM} 
= 11]?11)exten => *801,5,GotoIf($[${CALLERIDNUM} = 12]?12)exten => 
*801,6,GotoIf($[${CALLERIDNUM} = 13]?13)exten => 
*801,7,GotoIf($[${CALLERIDNUM} = 15]?15)exten => 
*801,8,Playback(beep)exten => *801,9,Hangupexten => 
*801,11,Goto(bjqueue,1,1)exten => *801,12,Goto(tonyqueue,1,1)exten 
=> *801,13,Goto(wendyqueue,1,1)exten => 
*801,15,Goto(danqueue,1,1)
 
exten 
=> *801,50,GotoIf($[${CALLERIDNUM} = 11]?61)exten => 
*801,51,GotoIf($[${CALLERIDNUM} = 12]?62)exten => 
*801,52,GotoIf($[${CALLERIDNUM} = 13]?63)exten => 
*801,53,GotoIf($[${CALLERIDNUM} = 15]?65)exten => 
*801,54,Playback(beep)exten => *801,55,Hangupexten => 
*801,61,Goto(bjqueue,2,1)exten => *801,62,Goto(tonyqueue,2,1)exten 
=> *801,63,Goto(wendyqueue,2,1)exten => 
*801,65,Goto(danqueue,2,1)
 
[bjqueue]exten => 1,1,Answerexten => 1,2,Wait(1)exten 
=> 1,3,Playback(agent-loginok)exten => 
1,4,SetGlobalVar(loggedin11=1)exten => 
1,5,AddQueueMember(tech,SIP/111)exten => 1,6,Hangup
 
exten 
=> 2,1,Answerexten => 2,2,Wait(1)exten => 
2,3,Playback(agent-loggedoff)exten => 
2,4,SetGlobalVar(loggedin11=0)exten => 
2,5,RemoveQueueMember(tech,SIP/111)exten => 
2,6,Hangup
 
The 
other users are setup pretty much the same as bjqueue.   I hope this 
helps.
 
B. 
J.
 
 
 
 


  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan 
  Stanton @ HomeSent: Monday, February 09, 2004 18:28To: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] Call 
  Queues
  Dear all,
  I am one of the people who answer the FWD welcome 
  line.  Since I don't want my phone ringing at 2am I have the 5 number 
  routed to a call queue.  Currently I have 2 extentions 271 which will log 
  my phone into the queue and 270 to log it out.  What I want to know is... 
  Is there a way to do this with just one exten?  I have tried with the 
  following :
   
  exten => 
  270,1,AddQueueMember(pulverWelcome)exten => 270,2,AGI(cepstral.agi,You 
  have just joined the Welcome queue)exten => 270,3,Hangupexten => 
  270,102,RemoveQueueMember(pulverWelcome)exten => 
  270,103,AGI(cepstral.agi,You have just left the Welcome queue)exten => 
  270,104,Hangup
  It works when I log in but when I try to log out 
  it crashes (403 error).  I know that the AddQueueMember returns -1, so 
  why does it not jump to priority 102 instead?
   
  Any ideas / sugestions 
  welcome.
   
   
  Jonathan


[Asterisk-Users] Call Queues

2004-02-09 Thread Jonathan Stanton @ Home



Dear all,
I am one of the people who answer the FWD welcome 
line.  Since I don't want my phone ringing at 2am I have the 5 number 
routed to a call queue.  Currently I have 2 extentions 271 which will log 
my phone into the queue and 270 to log it out.  What I want to know is... 
Is there a way to do this with just one exten?  I have tried with the 
following :
 
exten => 
270,1,AddQueueMember(pulverWelcome)exten => 270,2,AGI(cepstral.agi,You 
have just joined the Welcome queue)exten => 270,3,Hangupexten => 
270,102,RemoveQueueMember(pulverWelcome)exten => 
270,103,AGI(cepstral.agi,You have just left the Welcome queue)exten => 
270,104,Hangup
It works when I log in but when I try to log out it 
crashes (403 error).  I know that the AddQueueMember returns -1, so why 
does it not jump to priority 102 instead?
 
Any ideas / sugestions 
welcome.
 
 
Jonathan


smime.p7s
Description: S/MIME cryptographic signature


[Asterisk-Users] Call Queues

2004-02-01 Thread dkwok
I have setup AgentCallbackLogin and the agents have been logged in
successfully.
However when calls are queued and an agent picks up the call. It just
hang up the call.
On the command console it does say the agent "agent 1001 hang up on
customers. they must be pissed off". I agreed.
My queues.conf file:
[agents]
ackcall=no
agent => 1001,1001,xx ss
My queues.conf file:
[incoming]
announce = incoming
strategy=ringall
musice = default
member => Agent/1001
member => Agent/1002
My extensions.conf :

exten => 28,1,AgentCallbackLogin(|@local)
exten => 29,1,Queue(incoming)
In order to annonce to agent the correct queue does it have to have a
gsm file to playback the name of the queue ie "incoming" in this case?
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002


smime.p7s
Description: S/MIME Cryptographic Signature


[Asterisk-Users] Call Queues

2004-01-30 Thread dkwok
I have setup AgentCallbackLogin and the agents have been logged in 
successfully.

However when calls are queued and an agent picks up the call. It just 
hang up the call.

On the command console it does say the agent "agent 1001 hang up on 
customers. they must be pissed off". I agreed.

My queues.conf file:
[agents]
ackcall=no
agent => 1001,1001,xx ss
My queues.conf file:
[incoming]
announce = incoming
strategy=ringall
musice = default
member => Agent/1001
member => Agent/1002
My extensions.conf :

exten => 28,1,AgentCallbackLogin(|@local)
exten => 29,1,Queue(incoming)
In order to annonce to agent the correct queue does it have to have a 
gsm file to playback the name of the queue ie "incoming" in this case?

--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002


smime.p7s
Description: S/MIME Cryptographic Signature


[Asterisk-Users] Call Queues and Agents, using both devices and agent members

2003-06-30 Thread TC
Question for users of Agents and Queue.
In our queue.conf file we have entries for both agents and some
devices. We do this so that some dual pupose dispatchers can also
take a call if they have a spare moment.

We went live with a dispatch client over the weekend
a number of issues came up the most pressing is that when a call comes to
a member of the queue that is a device as opposed to an agent.
We randomly will get dead air when the uses takes the phone off hook to ans
the line.  The call does not get dropped from the queue, the user can just
hang up
and then the line will ring again and they can be online with the client
from the q.
This issue tends to happen more frequently if the dispatcher picks up the
line
on the first ring

Any idea's what might be causing this ??


A related issue
If we do a small test with 1 agent member and 1 device member
the system will try both the agent and the device if the agent does not have
a call on line already and the call goes to the agent,
the other device member of the q will ring
for the period of the timeout definition in the queue.conf, then if the
dispatcher
picks up the line they will also get a no call, Is this poosible to stop the
ring
on the line immediate after that call if ans by the agent ???




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call queues for phone operator

2003-06-14 Thread Mark Spencer
That's the right way.

MJark

On 13 Jun 2003, Brancaleoni Matteo wrote:

> Hi.
>
> I was wondering how can I make incoming calls to wait if the phone
> operator is busy. I've 8 incoming lines, with 30 extensions.
> What I need is if the operator is busy with call nr #1 , the new
> incoming call waits until the op. is free.
> Looking into app_queue seems the way to go.
> So I want to ask if I'm right or wrong:
> I set up only a queue , is to say operatorq, where
> the only member is my op. into technology SIP/operator
> So my queues.conf should be :
> [general]
> [default]
> [operatorq]
> music = default
> announce = from-queue
> ;context = qoutcon
> timeout = 20
> retry = 5
> ;maxlen = 0
> member => SIP/operator
>
> and when a call arrives, dial the operator and if he's busy,
> fire up app_queue .
>
> So what I expect, when the operator hangs up, his phone
> will automagically rings playing the announce "from-queue" and
> bridge it with the call that's waiting.
>
> So, I'm correct? Anyone experienced that or could give me
> a better way to handle that?
>
> Thanks a lot,
> matteo.
>
>
> --
> Brancaleoni Matteo <[EMAIL PROTECTED]>
> Espia - Emmegi Srl
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call queues for phone operator

2003-06-13 Thread Richard Lyman
http://www.digium.com/asterisk_handbook/agentlogin_queues.html

Brancaleoni Matteo wrote:
> 
> Hi.
> 
> I was wondering how can I make incoming calls to wait if the phone
> operator is busy. I've 8 incoming lines, with 30 extensions.
> What I need is if the operator is busy with call nr #1 , the new
> incoming call waits until the op. is free.
> Looking into app_queue seems the way to go.
> So I want to ask if I'm right or wrong:
> I set up only a queue , is to say operatorq, where
> the only member is my op. into technology SIP/operator
> So my queues.conf should be :
> [general]
> [default]
> [operatorq]
> music = default
> announce = from-queue
> ;context = qoutcon
> timeout = 20
> retry = 5
> ;maxlen = 0
> member => SIP/operator
> 
> and when a call arrives, dial the operator and if he's busy,
> fire up app_queue .
> 
> So what I expect, when the operator hangs up, his phone
> will automagically rings playing the announce "from-queue" and
> bridge it with the call that's waiting.
> 
> So, I'm correct? Anyone experienced that or could give me
> a better way to handle that?
> 
> Thanks a lot,
> matteo.
> 
> --
> Brancaleoni Matteo <[EMAIL PROTECTED]>
> Espia - Emmegi Srl
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call queues for phone operator

2003-06-13 Thread Brancaleoni Matteo
But I won't use agents.
I need only a queue for incoming calls to one extension,
that will answer and forward to other extens.
The operator won't be off hook all the time.

Matteo.


Il ven, 2003-06-13 alle 19:31, TC ha scritto:
> >Hi.
> >
> >I was wondering how can I make incoming calls to wait if the phone
> >operator is busy. I've 8 incoming lines, with 30 extensions.
> >What I need is if the operator is busy with call nr #1 , the new
> >incoming call waits until the op. is free.
> >Looking into app_queue seems the way to go.
> Thats correct
> >So I want to ask if I'm right or wrong:
> >and when a call arrives, dial the operator and if he's busy,
> >fire up app_queue .
> NO agents log into an agent q their phone is OFF-HOOK always
> thus if you Dial that agent ext it is always busy
> >So what I expect, when the operator hangs up, his phone
> >will automagically rings playing the announce "from-queue" and
> >bridge it with the call that's waiting.
> the agent will just hear beep beep & the optional announcement
> on the handset/speakerphone or headset
> then the inbound caller is bridged
> 
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call queues for phone operator

2003-06-13 Thread Martin Pycko
But he didn't think about agent. Just a regular SIP phone.
It should be in general like the original author of this thread thinks.
Besides it's easy to test so why not to test it :)

Martin

On Fri, 13 Jun 2003, TC wrote:

> >Hi.
> >
> >I was wondering how can I make incoming calls to wait if the phone
> >operator is busy. I've 8 incoming lines, with 30 extensions.
> >What I need is if the operator is busy with call nr #1 , the new
> >incoming call waits until the op. is free.
> >Looking into app_queue seems the way to go.
> Thats correct
> >So I want to ask if I'm right or wrong:
> >and when a call arrives, dial the operator and if he's busy,
> >fire up app_queue .
> NO agents log into an agent q their phone is OFF-HOOK always
> thus if you Dial that agent ext it is always busy
> >So what I expect, when the operator hangs up, his phone
> >will automagically rings playing the announce "from-queue" and
> >bridge it with the call that's waiting.
> the agent will just hear beep beep & the optional announcement
> on the handset/speakerphone or headset
> then the inbound caller is bridged
>
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call queues for phone operator

2003-06-13 Thread TC
>Hi.
>
>I was wondering how can I make incoming calls to wait if the phone
>operator is busy. I've 8 incoming lines, with 30 extensions.
>What I need is if the operator is busy with call nr #1 , the new
>incoming call waits until the op. is free.
>Looking into app_queue seems the way to go.
Thats correct
>So I want to ask if I'm right or wrong:
>and when a call arrives, dial the operator and if he's busy,
>fire up app_queue .
NO agents log into an agent q their phone is OFF-HOOK always
thus if you Dial that agent ext it is always busy
>So what I expect, when the operator hangs up, his phone
>will automagically rings playing the announce "from-queue" and
>bridge it with the call that's waiting.
the agent will just hear beep beep & the optional announcement
on the handset/speakerphone or headset
then the inbound caller is bridged



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Call queues for phone operator

2003-06-13 Thread Brancaleoni Matteo
Hi.

I was wondering how can I make incoming calls to wait if the phone
operator is busy. I've 8 incoming lines, with 30 extensions.
What I need is if the operator is busy with call nr #1 , the new
incoming call waits until the op. is free.
Looking into app_queue seems the way to go.
So I want to ask if I'm right or wrong:
I set up only a queue , is to say operatorq, where
the only member is my op. into technology SIP/operator
So my queues.conf should be :
[general]
[default]
[operatorq]
music = default
announce = from-queue
;context = qoutcon
timeout = 20
retry = 5
;maxlen = 0
member => SIP/operator

and when a call arrives, dial the operator and if he's busy,
fire up app_queue .

So what I expect, when the operator hangs up, his phone
will automagically rings playing the announce "from-queue" and
bridge it with the call that's waiting.

So, I'm correct? Anyone experienced that or could give me
a better way to handle that?

Thanks a lot, 
matteo.


-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users