Re: [asterisk-users] Feature request: What about a new DB_IFEXISTS function ?

[Satish Barot]

On Tue, Apr 9, 2013 at 7:56 PM, Olivier  wrote:

>
> Hello,
>
> What about having a new DB_IFEXISTS function in Asterisk ?
>
> It would accept two arguments : one being family/Key just as DB function,
> and the other being a default value which would be returned if family/key
> value does not exist.
>
> Thoughts ?
>
> Regards
>

You can achieve the same functionality using IF function. Something like,
...
same => n,Set(foo=${IF($[ "${DB(family/key)}" =
""]?defaultval:${DB(family/key)})})

--Satish Barot
Ahmedabad, India.
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Re: [asterisk-users] Feature request: What about a new DB_IFEXISTS function ?

[Olivier]

2013/4/10 Satish Barot 

>
> On Tue, Apr 9, 2013 at 7:56 PM, Olivier  wrote:
>
>>
>> Hello,
>>
>> What about having a new DB_IFEXISTS function in Asterisk ?
>>
>> It would accept two arguments : one being family/Key just as DB function,
>> and the other being a default value which would be returned if family/key
>> value does not exist.
>>
>> Thoughts ?
>>
>> Regards
>>
>
> You can achieve the same functionality using IF function. Something like,
> ...
> same => n,Set(foo=${IF($[ "${DB(family/key)}" =
> ""]?defaultval:${DB(family/key)})})
>

Yes, of course, but having a dedicated function for frequent pattern is
convenient.




>
> --Satish Barot
> Ahmedabad, India.
>
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Re: [asterisk-users] OT - How to simulate public IPs for lab testing

[Johan Wilfer]

Please, excuse me but I'm not sure I got your suggestion and  I'm
realizing I didn't correctly describe my lab set up.

At the moment, the router between both servers provides Internet access
to server1.
That means it has one WAN interface eth0 which is "on server2 side" and
one eth1 LAN interface which is on server1 side.
Currently, this router do NAT translation for server1.

Having clarified my setup, I guess your advice is to :
1. add address 1.2.3.4 to router's eth0
2. add address 4.3.2.1 to server2 interface
3. configure router to route trafic to 4.3.2.1 using server2 private
address (such as "ip route add 4.3.2.1/24  via
192.168.1.25")
4. configure server to route trafic to 1.2.3.4 using router private address

Is this correct ?



Sorry for the delay, but yes, that's how I do it.

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Re: [asterisk-users] my "blacklist" is not working

[Matthew Jordan]

On 04/09/2013 09:14 PM, Joseph wrote:
> On 04/09/13 19:27, Joseph wrote:
>>
>> exten => 4,n,Set(goaway=${CALLERID(number):0:2})
>> exten => 4,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1)
>> exten => 4,n,GotoIf($["${goaway}" = "V3" ]?blacklisted,s,1)
>>
>> But the first line suppose to compare the caller "ID" to the one in
>> database and jump to "blacklisted,s,1" context.
> 
> The reason it wasn't working as I had in database: "Manitoba" and the
> caller id this time was: "Manitoba " there was a space after the last
> letter, that is why it wasn't working. 
> Is there a way to counter/eliminate or do wildcard for "space"?
> 

Instead of doing an explicit comparison, you could always use a regular
expression to perform the matching:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_REGEX

Matt

-- 
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Digium, Inc. | Engineering Manager
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Re: [asterisk-users] my "blacklist" is not working

[Joseph]

On 04/10/13 08:29, Matthew Jordan wrote:

On 04/09/2013 09:14 PM, Joseph wrote:

On 04/09/13 19:27, Joseph wrote:


exten => 4,n,Set(goaway=${CALLERID(number):0:2})
exten => 4,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1)
exten => 4,n,GotoIf($["${goaway}" = "V3" ]?blacklisted,s,1)

But the first line suppose to compare the caller "ID" to the one in
database and jump to "blacklisted,s,1" context.


The reason it wasn't working as I had in database: "Manitoba" and the
caller id this time was: "Manitoba " there was a space after the last
letter, that is why it wasn't working.
Is there a way to counter/eliminate or do wildcard for "space"?



Instead of doing an explicit comparison, you could always use a regular
expression to perform the matching:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_REGEX

Matt

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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org


Thank you Matt for the pointer.
I need your help with it, how can I incorporate it into the dial plan?

In my case:
exten => 4,1,Set(goaway=${REGEX("[Manitora ]"  ${CALLERID(number))})
should return "1" if the caller id is "Manitoba" or "Manitoba "

What is next?

--
Joseph

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[asterisk-users] Logging SIP connection status for review

[Carlos Alvarez]

Is anyone using something to log SIP results (connected/not, latency) that
they really like?  We do some logging using simple scripts writing the
results of sip show peers to a text file if customers report issues, but it
would be nice to have a tool that logs all the time and lets us do some
better reporting.  For example, graphs of latency in a time range, or a
list of unreachable phones within a range, etc.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Logging SIP connection status for review

[Steve Edwards]

On Wed, 10 Apr 2013, Carlos Alvarez wrote:

Is anyone using something to log SIP results (connected/not, latency) 
that they really like?  We do some logging using simple scripts writing 
the results of sip show peers to a text file if customers report issues, 
but it would be nice to have a tool that logs all the time and lets us 
do some better reporting.  For example, graphs of latency in a time 
range, or a list of unreachable phones within a range, etc.


dumpcap can capture all of the SIP (and RTP) packets into a series of 
files without a huge performance hit.


A cron job can pbzip2 the files and delete if over x days old.

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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
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Re: [asterisk-users] Logging SIP connection status for review

[Carlos Alvarez]

On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards
wrote:

>
> dumpcap can capture all of the SIP (and RTP) packets into a series of
> files without a huge performance hit.
>
> A cron job can pbzip2 the files and delete if over x days old.


That's completely different.  We already run a good packet capture system.
 What I want to see is SIP registration statuses and latency logged about
once a minute.  We do that now by doing a 'sip show peers like x' and
putting it in a text file.  I can then correlate issues with times of high
latency or unreachable phones.  I'd just like to see more reporting and the
ability to correlate times and such.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Logging SIP connection status for review

[Ron Wheeler]

http://www.artifact-software.com/?page_id=1666
Would this help?
Put a JasperReport graph or two in a report step.


Ron

On 10/04/2013 2:02 PM, Steve Edwards wrote:

On Wed, 10 Apr 2013, Carlos Alvarez wrote:

Is anyone using something to log SIP results (connected/not, latency) 
that they really like?  We do some logging using simple scripts 
writing the results of sip show peers to a text file if customers 
report issues, but it would be nice to have a tool that logs all the 
time and lets us do some better reporting.  For example, graphs of 
latency in a time range, or a list of unreachable phones within a 
range, etc.


dumpcap can capture all of the SIP (and RTP) packets into a series of 
files without a huge performance hit.


A cron job can pbzip2 the files and delete if over x days old.



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

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[asterisk-users] ACD problem

[Tommy Cooper]

  Hi, 

I am working on a small inbound call center solution that uses an ACD system. I 
might add an IVR system later on. I only have 2 extensions set up (extensions 
1000 and 1001), I want the system to put new calls in a queue if both 
extensions are busy. I am currently subscribed with a SIP trunk provider and 
can successfully recieve calls. I want to design a system where customers can 
call my number, that call will then be directed to either extension 1000 or 
1001. If both extensions are in use, I want that 3rd call to be queued.

I don't think that the config below will direct calls to extension 1001 because 
the second line states that any incomming calls should be routed to extension 
1000. How do I change this so that calls are directed to all of my exensions?


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

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

agent => 1000,1000
agent => 1001,1001
 
When using the current config the caller will listen to the 'music on hold' 
until the agent answers but calls are only being forwarded to extension 1000 as 
stated above
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Re: [asterisk-users] ACD problem

[Paul Belanger]

On 13-04-10 04:08 PM, Tommy Cooper wrote:

   Hi,

I am working on a small inbound call center solution that uses an ACD system. I 
might add an IVR system later on. I only have 2 extensions set up (extensions 
1000 and 1001), I want the system to put new calls in a queue if both 
extensions are busy. I am currently subscribed with a SIP trunk provider and 
can successfully recieve calls. I want to design a system where customers can 
call my number, that call will then be directed to either extension 1000 or 
1001. If both extensions are in use, I want that 3rd call to be queued.

I don't think that the config below will direct calls to extension 1001 because 
the second line states that any incomming calls should be routed to extension 
1000. How do I change this so that calls are directed to all of my exensions?

Forget dialling the phones directly, let the queue deal with it. Dump 
everything in to the queue, then just wait for somebody to answer.




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

queues.conf

[general]
[support]

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

agent => 1000,1000
agent => 1001,1001

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



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Re: [asterisk-users] ACD problem

[Salman Zafar]

This line :
exten => *DID number*,2,Dial(SIP/1000)  is redundant and useless when you
are already using Queues. So just remove it and it should work.

What happen is, your dial-plan executes at 2nd priority DIAL a SIP
extension 1000 .. produce a call and at hang-up finishes no Queue/ACD
functionality is executed.




On Thu, Apr 11, 2013 at 1:08 AM, Tommy Cooper  wrote:

>   Hi,
>
> I am working on a small inbound call center solution that uses an ACD
> system. I might add an IVR system later on. I only have 2 extensions set up
> (extensions 1000 and 1001), I want the system to put new calls in a queue
> if both extensions are busy. I am currently subscribed with a SIP trunk
> provider and can successfully recieve calls. I want to design a system
> where customers can call my number, that call will then be directed to
> either extension 1000 or 1001. If both extensions are in use, I want that
> 3rd call to be queued.
> I don't think that the config below will direct calls to extension 1001
> because the second line states that any incomming calls should be routed to
> extension 1000. How do I change this so that calls are directed to all of
> my exensions?
>
> extensions.conf
> [from-myprovider]
> exten => *DID number*,1,Answer
> exten => *DID number*,2,Dial(SIP/1000)
> exten => *DID number*,3,Queue(support) ;not sure if this line belongs here
> exten => *DID number*,4,Hangup
>
> queues.conf
>
> [general]
> [support]
>
> musicclass=default
> strategy=rrmemory
> joinempty=no
> leavewhenempty=yes
> ringinuse=no
> Member => SIP/1000
> Member => SIP/1001
>
> agent => 1000,1000
> agent => 1001,1001
>
> When using the current config the caller will listen to the 'music on
> hold' until the agent answers but calls are only being forwarded to
> extension 1000 as stated above
>
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>



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Regards

**
Muhammad Salman
***
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[asterisk-users] AMI Reload action, returning generated errors?

[John Kiniston]

Howdy,

I'm building a webapp to allow my techs to do minor dialplan edits and
trigger a reload on my PBX's running 1.8

I have no problem triggering a 'reload pbx_config.so' via manager, The
problem is how can I see the results of my reload?

For example a missing close parenthesis which would show in
/var/log/asterisk/messages

[Apr 10 13:46:16] WARNING[23911] pbx_config.c: No closing parenthesis
found? 'Goto(7124,1' at line 39 of /etc/asterisk/extensions.conf

Or

[Apr 10 13:52:00] WARNING[24078] config.c: parse error: no closing ']',
line 1 of /etc/asterisk/extensions.conf


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build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
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Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?

[Daniel - Asterisk]

Hello Brynjolfur Thorvardsson,

Can I take a look at you CDR reporting tool?
I'm planning on using it on Postgresql but MySQL could be used too.

Thank you!

Elder D. Arohuanca
dCAP
Lima - Peru


On Fri, Feb 10, 2012 at 11:55 AM, asterisk jobs wrote:

> No, that doesn't do the job I specifically asked and installation
> instructions are all over the place...
>
> Thanks though.
>
>
> On Fri, Feb 10, 2012 at 11:36 AM, Tim Nelson wrote:
>
>> - Original Message -
>> >
>> > Yes, this is exactly what I am looking for - hopefully in English :-)
>> >
>> >
>> > Date or range selection would make this perfect. I have been looking
>> > for something like this for quite a while but there is none. I would
>> > really appreciate it if you share this with me.
>> >
>> >
>> > Question here, does the .php code read from database and displays or
>> > does it analyse the custom-cdr.csv file?
>> >
>> >
>>
>> Don't forget about the ever-popular Asterisk-stat and the newly revised
>> cdr-stats projects, both web based, proven, and work fantastic:
>>
>>
>> http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54
>> http://www.cdr-stats.org/
>>
>> --Tim
>>
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>>
>
>
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Re: [asterisk-users] ACD problem & outbound calls

[Tommy Cooper]

Thank you for your prompt reply. I removed that redundant line and now 
everything seems to work fine. Except outgoing calls that is, whenever i try to 
call an outside number the phone rings, the user can even answer back but then 
it hangs up after about 5 sec.
 
extensions.conf:
[sip-phone]
;This is the context setup for outgoing calls;exten => 
_NXXX.,2,Set(CALLERID(name)=*my number*)
;exten => _NXXX.,3,Set(CALLERID(num)=*my number*)
;exten => _NXX,1,Dial(mailto:SIP/$%7bexten...@myprovider.com)
 
;exten => _3.,1,Answer
exten => _3.,1,Dial(SIP/myprovider/${EXTEN:1},60) ;This is the only line 
that seems to work but the phone hangs up shortly after answering, as described 
above
;exten => _3.,5,Hangup
 
;exten => _X.,1,Answer
;exten => _X.,2,Set(CALLERID(name)=*my number*)
;exten => _X.,3,Set(CALLERID(num)=*my number*)
;exten => _X.,4,Dial(SIP/${EXTEN}@myprovider,30,Tt)
;exten => _X.,5,Hangup




From: Salman Zafar 
To: Tommy Cooper ; Asterisk Users Mailing List - 
Non-Commercial Discussion  
Sent: Wednesday, April 10, 2013 10:27 PM
Subject: Re: [asterisk-users] ACD problem


This line :
exten => *DID number*,2,Dial(SIP/1000)  is redundant and useless when you are 
already using Queues. So just remove it and it should work.

What happen is, your dial-plan executes at 2nd priority DIAL a SIP extension 
1000 .. produce a call and at hang-up finishes no Queue/ACD functionality is 
executed.






On Thu, Apr 11, 2013 at 1:08 AM, Tommy Cooper  wrote:

  Hi, 
>
>I am working on a small inbound call center solution that uses an ACD system. 
>I might add an IVR system later on. I only have 2 extensions set up 
>(extensions 1000 and 1001), I want the system to put new calls in a queue if 
>both extensions are busy. I am currently subscribed with a SIP trunk provider 
>and can successfully recieve calls. I want to design a system where customers 
>can call my number, that call will then be directed to either extension 1000 
>or 1001. If both extensions are in use, I want that 3rd call to be queued.
>
>I don't think that the config below will direct calls to extension 1001 
>because the second line states that any incomming calls should be routed to 
>extension 1000. How do I change this so that calls are directed to all of my 
>exensions?
>
>
>extensions.conf
>[from-myprovider]
>exten => *DID number*,1,Answer
>exten => *DID number*,2,Dial(SIP/1000)
>exten => *DID number*,3,Queue(support) ;not sure if this line belongs here
>exten => *DID number*,4,Hangup
>
>queues.conf
>
>[general]
>[support]
>
>musicclass=default
>strategy=rrmemory
>joinempty=no
>leavewhenempty=yes
>ringinuse=no
>Member => SIP/1000
>Member => SIP/1001
>
>agent => 1000,1000
>agent => 1001,1001
>
>When using the current config the caller will listen to the 'music on hold' 
>until the agent answers but calls are only being forwarded to extension 1000 
>as stated above
>
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**
Muhammad Salman
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[asterisk-users] Follow Me CID

[Nyamul Hassan]

Hi,

We have recently implemented asterisk as the office PBX, and as a GUI, we
chose Elastix.  All seems quite easy and am very pleased with the way
Asterisk is running and handling calls.  To summarize briefly, our setup
looks like this:

LAN -> Asterisk PBX (Elastix) -> Asterisk TDM (basic CLI with libSS7)

We get a whole 3 digit number series from our telco.  So, like, +1 312 765
7xxx (just an example), where we get to choose XXX on our own.  To simplify
things, we have forwarded the full incoming extension to "Asterisk PBX",
and have assigned full numbered SIP IDs to our users.  And so far, our
users can send / receive calls without any problem!

The problem starts when we are trying out the "Follow Me" feature, and
there seems to be a problem with Caller ID.  The following scenario works:

Internal User --- > [calling] > Internal User --- > [call failing] >
Forwarded number in the settings of Internal User

This works like a charm!  However, the problem arises when, the initial
caller is an external user.  After looking at the verbose debug messages
for a while, we found that core issue is Caller ID.  Our outbound Telco
will not accept calls from anything that does not start with 312 765 7xxx.

Is there a way to change the caller ID of outbound calls to the "dialed
internal extension CallerID"?

We also came up with another link on the internet, where apparently there
are settings on the SS7 related to this type of setup.  Probably RDNIS and
OCN are the ones.  Is there a way to make "Asterisk PBX" ask "Asterisk TDM"
to make a forwarded calls, and then subsequently pass those settings to the
Telco?

Regards
HASSAN
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[asterisk-users] Setting a CDR field from using feature codes...

[Carlos Chavez]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I am trying to set the CDR(userfield) to a certain vaule using the
application map of features.conf but I am not able to do it.  When I
receive a call I would like to tag it with a client code (3 digit
numeric) so I can referenci it later from the CDR.  I have edited
features.conf with something like:

code => #111,self,SET(CDR(userfield(111))

or

code => #111,self,AGI(code.agi)

The DYNAMIC_FEATURES variable is in the globals section and includes
the application map name.  When I do a "features reload" I can see
everything loads and when I dial the code during a call I can see a
message like:

- --  Feature Found: code exten: code

The problem is that my CDR variable is not being written to.  The
first example does not show anything on screen.  For the second when I
turn agi debug on I can see:

AGI Rx << SET VARIABLE CDR(userfield) "111"

But when I hang up neither my h extension or the CDR itself will show
the value I set, it is empty.  I do not know what I am doing wrong or
maybe CDR variables are not available from features?

- -- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Logging SIP connection status for review

[Duncan Turnbull]

On On Wed, 10 Apr 2013, Carlos Alvarez wrote: 
>> 
>>> Is anyone using something to log SIP results (connected/not, latency) that 
>>> they really like?  We do some logging using simple scripts writing the 
>>> results of sip show peers to a text file if customers report issues, but it 
>>> would be nice to have a tool that logs all the time and lets us do some 
>>> better reporting.  For example, graphs of latency in a time range, or a 
>>> list of unreachable phones within a range, etc. 
>> 

How about munin

http://munin-monitoring.org/

With these plugins
http://www.venturevoip.com/news.php?rssid=2322

You could easily adapt one to do registrations. I find the sip peers and calls 
in the last hour quite interesting

Cheers Duncan

>> 
>> 
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> phone: 866-970-2435, ext 102
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Re: [asterisk-users] AMI Reload action, returning generated errors?

[Trung Nguyen Dac]

*Hi pal.*
*
*
*Maybe you forgot to enclose the near  parenthesis found? 'Goto(7124,1' in
extension file, Firstly correct this syntax then try again.*
*
*
*BRs*

*/***
*Nguyen Trung (Mr)*
*Trust me I'm an "Engineer"!*
*   *
*   VEGA CORPORATION
  *
*

   - Em@il: trun...@vega.com.vn
   - Handphone: [+84] 934448537
   - Skype: nguyentrunght

*/
**

*


On Thu, Apr 11, 2013 at 3:59 AM, John Kiniston wrote:

> Howdy,
>
> I'm building a webapp to allow my techs to do minor dialplan edits and
> trigger a reload on my PBX's running 1.8
>
> I have no problem triggering a 'reload pbx_config.so' via manager, The
> problem is how can I see the results of my reload?
>
> For example a missing close parenthesis which would show in
> /var/log/asterisk/messages
>
> [Apr 10 13:46:16] WARNING[23911] pbx_config.c: No closing parenthesis
> found? 'Goto(7124,1' at line 39 of /etc/asterisk/extensions.conf
>
> Or
>
> [Apr 10 13:52:00] WARNING[24078] config.c: parse error: no closing ']',
> line 1 of /etc/asterisk/extensions.conf
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
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Re: [asterisk-users] ACD problem

[Bharat Lalcheta]

Hi,

You can check extension status using chanisavail function. And extension is
not free, you can divert your call to queue.


http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail

Regards,

Bharat Lalcheta

On Thu, Apr 11, 2013 at 1:38 AM, Tommy Cooper  wrote:

>   Hi,
>
> I am working on a small inbound call center solution that uses an ACD
> system. I might add an IVR system later on. I only have 2 extensions set up
> (extensions 1000 and 1001), I want the system to put new calls in a queue
> if both extensions are busy. I am currently subscribed with a SIP trunk
> provider and can successfully recieve calls. I want to design a system
> where customers can call my number, that call will then be directed to
> either extension 1000 or 1001. If both extensions are in use, I want that
> 3rd call to be queued.
> I don't think that the config below will direct calls to extension 1001
> because the second line states that any incomming calls should be routed to
> extension 1000. How do I change this so that calls are directed to all of
> my exensions?
>
> extensions.conf
> [from-myprovider]
> exten => *DID number*,1,Answer
> exten => *DID number*,2,Dial(SIP/1000)
> exten => *DID number*,3,Queue(support) ;not sure if this line belongs here
> exten => *DID number*,4,Hangup
>
> queues.conf
>
> [general]
> [support]
>
> musicclass=default
> strategy=rrmemory
> joinempty=no
> leavewhenempty=yes
> ringinuse=no
> Member => SIP/1000
> Member => SIP/1001
>
> agent => 1000,1000
> agent => 1001,1001
>
> When using the current config the caller will listen to the 'music on
> hold' until the agent answers but calls are only being forwarded to
> extension 1000 as stated above
>
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Re: [asterisk-users] ACD problem

[Lenz Emilitri]

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


2013/4/10 Tommy Cooper 

>   Hi,
>
> I am working on a small inbound call center solution that uses an ACD
> system. I might add an IVR system later on. I only have 2 extensions set up
> (extensions 1000 and 1001), I want the system to put new calls in a queue
> if both extensions are busy. I am currently subscribed with a SIP trunk
> provider and can successfully recieve calls. I want to design a system
> where customers can call my number, that call will then be directed to
> either extension 1000 or 1001. If both extensions are in use, I want that
> 3rd call to be queued.
> I don't think that the config below will direct calls to extension 1001
> because the second line states that any incomming calls should be routed to
> extension 1000. How do I change this so that calls are directed to all of
> my exensions?
>
> extensions.conf
> [from-myprovider]
> exten => *DID number*,1,Answer
> exten => *DID number*,2,Dial(SIP/1000)
> exten => *DID number*,3,Queue(support) ;not sure if this line belongs here
> exten => *DID number*,4,Hangup
>
> queues.conf
>
> [general]
> [support]
>
> musicclass=default
> strategy=rrmemory
> joinempty=no
> leavewhenempty=yes
> ringinuse=no
> Member => SIP/1000
> Member => SIP/1001
>
> agent => 1000,1000
> agent => 1001,1001
>
> When using the current config the caller will listen to the 'music on
> hold' until the agent answers but calls are only being forwarded to
> extension 1000 as stated above
>
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