Re: [asterisk-users] Asterisk 11 -CDR values changed in hangup handler not saved ?

2013-03-29 Thread Julian Lyndon-Smith
check out the endbeforehexten option in cdr.conf

this needs to set to yes

Julian


On 28 March 2013 23:56, Olivier oza_4...@yahoo.fr wrote:

 Hello,

 I'm using Hanhup Handlers in a testing asterisk 11 system.
 Within one such handler, I'm setting CDR values.

 To me, it seems those changed CDR values are not saved in CDR back-end.

 Can you confirm ?

 Regards

 --
 _
 -- 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




-- 
Julian Lyndon-Smith
IT Director, Dot R Limited

I don’t care if it works on your machine!  We are not shipping your
machine!”

The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
--
_
-- 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] Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]

2013-03-29 Thread Olivier
2013/3/29 Julian Lyndon-Smith aster...@dotr.com

 check out the endbeforehexten option in cdr.conf

 this needs to set to yes

 Julian



Unfortunately, this doesn't help.

Let's drop the hangup handler at the moment, and focus on the saving to
file part.
Then my issue is I can't update CDR value is hangup exten.

Here is a dialplan that illustrate this:

[from-foobar]
exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel
${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and
${CALLERID(num)})
  same = n, Set(CDR(userfield)=foo)
  same = n, Dial(SIP/foobar/${EXTEN})
  same = n, Set(CDR(userfield)=bar)
  same = n, Hangup()

exten = h,1,Verbose(0,Entering context ${CONTEXT} from
${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to
${EXTEN} and ${CALLERID(num)})
  same = n,
ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2)

My goal is to get either baz1 or baz2 value in
/var/log/asterisk/cdr-csv/Master.csv.

Typing channel originate Local/7005@from-foobar application Playback
tt-monkeys, I can see that the line with ExecIf is run but CDR still
contains foo value (the one set before Dial).
The strange thing is :
1. a CDR is written at the moment extension 7005 answers,
2. no other CDR is added when 7005 hangs up (so can't tell how long
extension 7005 listened to monkeys fellows).
(Setting endbeforehexten to either yes or no has no effect on this
behaviour.


My question are:
1. Is it simply possible to update CDR in hangup exten ?
2. How can I have a CDR for the application Playback part (see above) ?
3. Any tip or suggestion ?

Cheers
--
_
-- 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] Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]

2013-03-29 Thread Julian Lyndon-Smith
Ah, right. Have a look at this documentation:

You may need to add some mapping

Julian
cdr_custom

This CDR backend allows for custom formatting of CDR records in a log file.
This module is most commonly used for customized CSV output. The
configuration file used for this module is /etc/asterisk/cdr_custom.conf. A
single section called [mappings] should exist in this file. The
[mappings] section
contains mappings between a filename and the custom template for a CDR. The
template is specified using Asterisk dialplan functions.

The following example shows a sample configuration for cdr_custom that
enables a single CDR log file, Master.csv. This file will be created as
/var/log/asterisk/cdr-custom/Master.csv. The template that has been defined
uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() function
retrieves values from the CDR being logged. The CSV_QUOTE() function
ensures that the values are properly escaped for the CSV file format:

[mappings]

Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},
   ${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},
   ${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},
   ${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},
   ${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},
   ${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},
   ${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},
   ${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},
   ${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})}

In the actual configuration file, the value in the Master.csv mapping
should be on a single line.
cdr_manager


On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote:




 2013/3/29 Julian Lyndon-Smith aster...@dotr.com

 check out the endbeforehexten option in cdr.conf

 this needs to set to yes

 Julian



 Unfortunately, this doesn't help.

 Let's drop the hangup handler at the moment, and focus on the saving to
 file part.
 Then my issue is I can't update CDR value is hangup exten.

 Here is a dialplan that illustrate this:

 [from-foobar]
 exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel
 ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and
 ${CALLERID(num)})
   same = n, Set(CDR(userfield)=foo)
   same = n, Dial(SIP/foobar/${EXTEN})
   same = n, Set(CDR(userfield)=bar)
   same = n, Hangup()

 exten = h,1,Verbose(0,Entering context ${CONTEXT} from
 ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to
 ${EXTEN} and ${CALLERID(num)})
   same = n,
 ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2)

 My goal is to get either baz1 or baz2 value in
 /var/log/asterisk/cdr-csv/Master.csv.

 Typing channel originate Local/7005@from-foobar application Playback
 tt-monkeys, I can see that the line with ExecIf is run but CDR still
 contains foo value (the one set before Dial).
 The strange thing is :
 1. a CDR is written at the moment extension 7005 answers,
 2. no other CDR is added when 7005 hangs up (so can't tell how long
 extension 7005 listened to monkeys fellows).
 (Setting endbeforehexten to either yes or no has no effect on this
 behaviour.


 My question are:
 1. Is it simply possible to update CDR in hangup exten ?
 2. How can I have a CDR for the application Playback part (see above) ?
 3. Any tip or suggestion ?

 Cheers

 --
 _
 -- 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




-- 
Julian Lyndon-Smith
IT Director, Dot R Limited

I don’t care if it works on your machine!  We are not shipping your
machine!”

The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
--
_
-- 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] sip set debug on output to file only (not to console)

2013-03-29 Thread Marie Fischer
Hello everybody,

I am trying to find an intermittent SIP error with one provider and thought the 
best first step would be to have sip set debug on for some days and check the 
logs.

Everything gets logged nicely, but the SIP log clutters up the console quite 
badly. Is it possible to have SIP debug log go only to the log file and not to 
the console?

My logger.conf:

console = notice,warning,error
messages = notice,warning,error
full = notice,warning,error,debug,verbose,dtmf,fax

On the console, I entered:

core set verbose 3
core set debug 0
sip set debug on

Thanks,

-- 

marie




--
_
-- 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] sip set debug on output to file only (not to console)

2013-03-29 Thread Doug Lytle

Marie Fischer wrote:

full = notice,warning,error,debug,verbose,dtmf,fax


You should have a log called full in:

/var/log/asterisk

Doug


--
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


--
_
-- 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] sip set debug on output to file only (not to console)

2013-03-29 Thread Doug Lytle

Marie Fischer wrote:

but the SIP log clutters up the console quite badly


I guess I should slow down when reading.  Sorry for the noise.

Doug


--
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


--
_
-- 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] sip set debug on output to file only (not to console)

2013-03-29 Thread Marie Fischer
On 29.03.2013, at 15:05, Doug Lytle supp...@drdos.info wrote:

 Marie Fischer wrote:
 full = notice,warning,error,debug,verbose,dtmf,fax
 
 You should have a log called full in:
 
 /var/log/asterisk

Sure I do and happy with that. :)

The point is, I also have my Asterisk console full of SIP messages and I asked 
if it was possible to switch those off.

-- 

marie


--
_
-- 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] Asterisk 11 -CDR values changed in hangup handler not saved ?

2013-03-29 Thread Mitch Claborn

I have seen that behavior also.


Mitch

On 03/28/2013 06:56 PM, Olivier wrote:

Hello,

I'm using Hanhup Handlers in a testing asterisk 11 system.
Within one such handler, I'm setting CDR values.

To me, it seems those changed CDR values are not saved in CDR back-end.

Can you confirm ?

Regards


--
_
-- 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


Re: [asterisk-users] sip set debug on output to file only (not to console)

2013-03-29 Thread Asghar Mohammad
hi,
open debug only on problematic peer.
sip set debug peer peer name
or
sip set debug ip peer ip


On Fri, Mar 29, 2013 at 2:02 PM, Marie Fischer ma...@vtl.ee wrote:

 Hello everybody,

 I am trying to find an intermittent SIP error with one provider and
 thought the best first step would be to have sip set debug on for some
 days and check the logs.

 Everything gets logged nicely, but the SIP log clutters up the console
 quite badly. Is it possible to have SIP debug log go only to the log file
 and not to the console?

 My logger.conf:

 console = notice,warning,error
 messages = notice,warning,error
 full = notice,warning,error,debug,verbose,dtmf,fax

 On the console, I entered:

 core set verbose 3
 core set debug 0
 sip set debug on

 Thanks,

 --

 marie




 --
 _
 -- 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

Re: [asterisk-users] sip set debug on output to file only (not to console)

2013-03-29 Thread Mitch Claborn
I recently faced the same issue.  I didn't find a way in Asterisk to do 
what I wanted.


A good workaround is to use wireshark in batch mode (tshark) to trace 
traffic to the IP address you are interested in.  You should be able to 
filter it to capture only SIP traffic.



Mitch

On 03/29/2013 08:02 AM, Marie Fischer wrote:

Hello everybody,

I am trying to find an intermittent SIP error with one provider and thought the best 
first step would be to have sip set debug on for some days and check the logs.

Everything gets logged nicely, but the SIP log clutters up the console quite 
badly. Is it possible to have SIP debug log go only to the log file and not to 
the console?

My logger.conf:

console = notice,warning,error
messages = notice,warning,error
full = notice,warning,error,debug,verbose,dtmf,fax

On the console, I entered:

core set verbose 3
core set debug 0
sip set debug on

Thanks,



--
_
-- 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] Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]

2013-03-29 Thread Olivier
Thanks but I willingly choose a standard CDR field (I checked with both
accountcode and userfield) which appears in
/var/log/asterisk/cdr-csv/Master.csv (to keep cdr-cusdom/Master.csv away to
simplify things)
the fact found in Master.csv is  foo, the value set before entering the
hangup extension (see previous dialplan)).

To me, this is either a feature (you can't set CDR values in hangup
exten) or a bug.

How would you qualify this ?


2013/3/29 Julian Lyndon-Smith aster...@dotr.com

 Ah, right. Have a look at this documentation:

 You may need to add some mapping

 Julian
 cdr_custom

 This CDR backend allows for custom formatting of CDR records in a log
 file. This module is most commonly used for customized CSV output. The
 configuration file used for this module is /etc/asterisk/cdr_custom.conf.
 A single section called [mappings] should exist in this file. The
 [mappings] section contains mappings between a filename and the custom
 template for a CDR. The template is specified using Asterisk dialplan
 functions.

 The following example shows a sample configuration for cdr_custom that
 enables a single CDR log file, Master.csv. This file will be created as
 /var/log/asterisk/cdr-custom/Master.csv. The template that has been
 defined uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() 
 function
 retrieves values from the CDR being logged. The CSV_QUOTE() function
 ensures that the values are properly escaped for the CSV file format:

 [mappings]

 Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},
${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},
${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},
${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},
${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},
${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},
${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},
${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},
${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})}

 In the actual configuration file, the value in the Master.csv mapping
 should be on a single line.
 cdr_manager


 On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote:




 2013/3/29 Julian Lyndon-Smith aster...@dotr.com

 check out the endbeforehexten option in cdr.conf

 this needs to set to yes

 Julian



 Unfortunately, this doesn't help.

 Let's drop the hangup handler at the moment, and focus on the saving to
 file part.
 Then my issue is I can't update CDR value is hangup exten.

  Here is a dialplan that illustrate this:

 [from-foobar]
 exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel
 ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and
 ${CALLERID(num)})
   same = n, Set(CDR(userfield)=foo)
   same = n, Dial(SIP/foobar/${EXTEN})
   same = n, Set(CDR(userfield)=bar)
   same = n, Hangup()

 exten = h,1,Verbose(0,Entering context ${CONTEXT} from
 ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to
 ${EXTEN} and ${CALLERID(num)})
   same = n,
 ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2)

 My goal is to get either baz1 or baz2 value in
 /var/log/asterisk/cdr-csv/Master.csv.

 Typing channel originate Local/7005@from-foobar application Playback
 tt-monkeys, I can see that the line with ExecIf is run but CDR still
 contains foo value (the one set before Dial).
 The strange thing is :
 1. a CDR is written at the moment extension 7005 answers,
 2. no other CDR is added when 7005 hangs up (so can't tell how long
 extension 7005 listened to monkeys fellows).
 (Setting endbeforehexten to either yes or no has no effect on this
 behaviour.


 My question are:
 1. Is it simply possible to update CDR in hangup exten ?
 2. How can I have a CDR for the application Playback part (see above) ?
 3. Any tip or suggestion ?

 Cheers

 --
 _
 -- 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




 --
 Julian Lyndon-Smith
 IT Director, Dot R Limited

 I don’t care if it works on your machine!  We are not shipping your
 machine!”

 The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg

 --
 _
 -- 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] To queue or not to queue...

2013-03-29 Thread Lenz Emilitri
Hello Gregory,
I wouldn't say this is a typical scenario for using a ringall queue,
especially if the agent set gets larger and larger. On the other side, a
ringgroup won't solve the issue of ringing all those phones at once.  What
I would be looking into, considered the motivation of your agents, is to
split the system into more than one queue and send the calls randomly to
each queue. If everybody is busy you get out and retry. This should not
impact call answer times as long as you have 30/40 people available per
queue - but your box will handle a fraction of the load and you can easily
partition such a system on multiple boxes.
Just my two cents,
l.


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

 Hello All,

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

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

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

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

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

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


 --
 _
 -- 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




-- 
Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com
--
_
-- 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] Asterisk 11 -CDR values changed in hangup handler not saved ?

2013-03-29 Thread Olivier
How would you qualify it ?
A feature ? A bug ?

Could you find a work around ?



2013/3/29 Mitch Claborn mitch...@claborn.net

 I have seen that behavior also.


 Mitch


 On 03/28/2013 06:56 PM, Olivier wrote:

 Hello,

 I'm using Hanhup Handlers in a testing asterisk 11 system.
 Within one such handler, I'm setting CDR values.

 To me, it seems those changed CDR values are not saved in CDR back-end.

 Can you confirm ?

 Regards


 --
 __**__**_
 -- 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-**usershttp://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-**usershttp://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] Getting Unknown Error while configuring Asterisk with Linux HA

2013-03-29 Thread Ahmed Munir
Hi,

I recently configured Linux HA for Asterisk service (using Asterisk
resource agent downloaded from link:
https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk
).
As per configuration it is working good but when I include monitor_sipuri=
sip:42@10.3.152.103  parameter in primitive section it is giving me an
errors like listed below;

root@asterisk2 ~ crm_mon -1



Last updated: Thu Mar 28 06:09:54 2013

Stack: Heartbeat

Current DC: asterisk2 (b966dfa2-5973-4dfc-96ba-b2d38319c174) - partition
with quorum

Version: 1.0.12-unknown

2 Nodes configured, unknown expected votes

1 Resources configured.





Online: [ asterisk1 asterisk2 ]



Resource Group: group_1

 asterisk_2 (lsb:asterisk): Started asterisk1

 IPaddr_10_3_152_103(ocf::heartbeat:IPaddr):Started
asterisk1



Failed actions:

p_asterisk_start_0 (node=asterisk1, call=64, rc=1, status=complete):
unknown error

p_asterisk_start_0 (node=asterisk2, call=20, rc=1, status=complete):
unknown error


I tested the 'sipsak' tool on cli, it is executing without any issue i.e.
returning 200 OK but when I remove this param monitor_sipuri I'm not
getting the errors and also I created sip profile '42' without setting any
password, tested first on softphone and is working.

Test result for sipsak;

root@asterisk1 ~ sipsak -v -s sip:42@10.3.152.103
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.152.105:60928
;branch=z9hG4bK.274e15e9;alias;received=10.3.152.103;rport=60928
From: sip:sipsak@10.3.152.105:60928;tag=68c5c65d
To: sip:42@10.3.152.103;tag=as558d9271
Call-ID: 1757791837@10.3.152.105
CSeq: 1 OPTIONS
Server: Asterisk PBX 10.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: sip:10.3.152.103:5060
Accept: application/sdp
Content-Length: 0


Listing down the configuration below which I configured;

 node $id=887bae58-1eb6-47d1-b539-d12a2ed3d836 asterisk1
node $id=b966dfa2-5973-4dfc-96ba-b2d38319c174 asterisk2
primitive IPaddr_10_3_152_103 ocf:heartbeat:IPaddr \
op monitor interval=5s timeout=20s \
params ip=10.3.152.103
primitive p_asterisk ocf:heartbeat:asterisk \
op monitor interval=10s \
params realtime=true
group group_1 p_asterisk IPaddr_10_3_152_103 \
meta target-role=Started
location rsc_location_group_1 group_1 \
rule $id=preferred_location_group_1 100: #uname eq asterisk1
colocation asterisk-with-ip inf: p_asterisk IPaddr_10_3_152_103
property $id=cib-bootstrap-options \
symmetric-cluster=true \
no-quorum-policy=stop \
default-resource-stickiness=0 \
stonith-enabled=false \
stonith-action=reboot \
startup-fencing=true \
stop-orphan-resources=true \
stop-orphan-actions=true \
remove-after-stop=false \
default-action-timeout=120s \
is-managed-default=true \
cluster-delay=60s \
pe-error-series-max=-1 \
pe-warn-series-max=-1 \
pe-input-series-max=-1 \
dc-version=1.0.12-unknown \
cluster-infrastructure=Heartbeat

And the status I'm getting is listed below;

root@asterisk1 ~ crm_mon -1

Last updated: Fri Mar 29 12:25:10 2013
Stack: Heartbeat
Current DC: asterisk1 (887bae58-1eb6-47d1-b539-d12a2ed3d836) - partition
with quorum
Version: 1.0.12-unknown
2 Nodes configured, unknown expected votes
1 Resources configured.


Online: [ asterisk1 asterisk2 ]

 Resource Group: group_1
 p_asterisk (ocf::heartbeat:asterisk):  Started asterisk1
 IPaddr_10_3_152_103(ocf::heartbeat:IPaddr):Started
asterisk1


Please advise to overcome this issue.


-- 
Regards,

Ahmed Munir Chohan
--
_
-- 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] Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]

2013-03-29 Thread Julian Lyndon-Smith
you can't set values in the h extension _unless_ you use the
endbeforehexten option in cdr.conf

you may need to reload the cdr module or restart asterisk for the
option to take effect.

It works. I know it does, as I use hangup handlers all the time. Much
better than the h extension ;)


Julian

On 29 March 2013 14:06, Olivier oza_4...@yahoo.fr wrote:
 Thanks but I willingly choose a standard CDR field (I checked with both
 accountcode and userfield) which appears in
 /var/log/asterisk/cdr-csv/Master.csv (to keep cdr-cusdom/Master.csv away to
 simplify things)
 the fact found in Master.csv is  foo, the value set before entering the
 hangup extension (see previous dialplan)).

 To me, this is either a feature (you can't set CDR values in hangup exten)
 or a bug.

 How would you qualify this ?


 2013/3/29 Julian Lyndon-Smith aster...@dotr.com

 Ah, right. Have a look at this documentation:

 You may need to add some mapping

 Julian

 cdr_custom

 This CDR backend allows for custom formatting of CDR records in a log
 file. This module is most commonly used for customized CSV output. The
 configuration file used for this module is /etc/asterisk/cdr_custom.conf. A
 single section called [mappings] should exist in this file. The [mappings]
 section contains mappings between a filename and the custom template for a
 CDR. The template is specified using Asterisk dialplan functions.

 The following example shows a sample configuration for cdr_custom that
 enables a single CDR log file, Master.csv. This file will be created as
 /var/log/asterisk/cdr-custom/Master.csv. The template that has been defined
 uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() function
 retrieves values from the CDR being logged. The CSV_QUOTE() function ensures
 that the values are properly escaped for the CSV file format:

 [mappings]

 Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},
${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},
${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},
${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},
${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},
${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},
${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},
${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},
${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})}

 In the actual configuration file, the value in the Master.csv mapping
 should be on a single line.

 cdr_manager



 On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote:




 2013/3/29 Julian Lyndon-Smith aster...@dotr.com

 check out the endbeforehexten option in cdr.conf

 this needs to set to yes

 Julian



 Unfortunately, this doesn't help.

 Let's drop the hangup handler at the moment, and focus on the saving to
 file part.
 Then my issue is I can't update CDR value is hangup exten.

 Here is a dialplan that illustrate this:

 [from-foobar]
 exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel
 ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and
 ${CALLERID(num)})
   same = n, Set(CDR(userfield)=foo)
   same = n, Dial(SIP/foobar/${EXTEN})
   same = n, Set(CDR(userfield)=bar)
   same = n, Hangup()

 exten = h,1,Verbose(0,Entering context ${CONTEXT} from
 ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to
 ${EXTEN} and ${CALLERID(num)})
   same = n,
 ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2)

 My goal is to get either baz1 or baz2 value in
 /var/log/asterisk/cdr-csv/Master.csv.

 Typing channel originate Local/7005@from-foobar application Playback
 tt-monkeys, I can see that the line with ExecIf is run but CDR still
 contains foo value (the one set before Dial).
 The strange thing is :
 1. a CDR is written at the moment extension 7005 answers,
 2. no other CDR is added when 7005 hangs up (so can't tell how long
 extension 7005 listened to monkeys fellows).
 (Setting endbeforehexten to either yes or no has no effect on this
 behaviour.


 My question are:
 1. Is it simply possible to update CDR in hangup exten ?
 2. How can I have a CDR for the application Playback part (see above) ?
 3. Any tip or suggestion ?

 Cheers

 --
 _
 -- 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




 --
 Julian Lyndon-Smith
 IT Director, Dot R Limited

 I don’t care if it works on your machine!  We are not shipping your
 machine!”

 The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg

 --
 _
 -- Bandwidth and Colocation 

Re: [asterisk-users] Asterisk 11 -CDR values changed in hangup handler not saved ?

2013-03-29 Thread Mitch Claborn
My personal opinion is that it is a design flaw. It is probably working 
as designed, but I think the design should be different.


I did not find any workaround.


Mitch

On 03/29/2013 11:14 AM, Olivier wrote:

How would you qualify it ?
A feature ? A bug ?

Could you find a work around ?



2013/3/29 Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net

I have seen that behavior also.


Mitch


On 03/28/2013 06:56 PM, Olivier wrote:

Hello,

I'm using Hanhup Handlers in a testing asterisk 11 system.
Within one such handler, I'm setting CDR values.

To me, it seems those changed CDR values are not saved in CDR
back-end.

Can you confirm ?

Regards


--

_
-- 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
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
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



--
_
-- 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] Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]

2013-03-29 Thread Olivier
Le 29 mars 2013 18:26, Julian Lyndon-Smith aster...@dotr.com a écrit :

 you can't set values in the h extension _unless_ you use the
 endbeforehexten option in cdr.conf
I did  and  couldn't get it to work  :-(

 you may need to reload the cdr module or restart asterisk for the
 option to take effect.

 It works.
You mean  it does work in asterisk 11 ?
Which CDR variables did you then play with ?

I know it does, as I use hangup handlers all the time. Much
 better than the h extension ;)
I fully agree !



 Julian

 On 29 March 2013 14:06, Olivier oza_4...@yahoo.fr wrote:
  Thanks but I willingly choose a standard CDR field (I checked with both
  accountcode and userfield) which appears in
  /var/log/asterisk/cdr-csv/Master.csv (to keep cdr-cusdom/Master.csv
away to
  simplify things)
  the fact found in Master.csv is  foo, the value set before entering the
  hangup extension (see previous dialplan)).
 
  To me, this is either a feature (you can't set CDR values in hangup
exten)
  or a bug.
 
  How would you qualify this ?
 
 
  2013/3/29 Julian Lyndon-Smith aster...@dotr.com
 
  Ah, right. Have a look at this documentation:
 
  You may need to add some mapping
 
  Julian
 
  cdr_custom
 
  This CDR backend allows for custom formatting of CDR records in a log
  file. This module is most commonly used for customized CSV output. The
  configuration file used for this module is
/etc/asterisk/cdr_custom.conf. A
  single section called [mappings] should exist in this file. The
[mappings]
  section contains mappings between a filename and the custom template
for a
  CDR. The template is specified using Asterisk dialplan functions.
 
  The following example shows a sample configuration for cdr_custom that
  enables a single CDR log file, Master.csv. This file will be created as
  /var/log/asterisk/cdr-custom/Master.csv. The template that has been
defined
  uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR()
function
  retrieves values from the CDR being logged. The CSV_QUOTE() function
ensures
  that the values are properly escaped for the CSV file format:
 
  [mappings]
 
  Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},
 ${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},
 ${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},
 ${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},
 ${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},
 ${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},
 ${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},
 ${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},
 ${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})}
 
  In the actual configuration file, the value in the Master.csv mapping
  should be on a single line.
 
  cdr_manager
 
 
 
  On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote:
 
 
 
 
  2013/3/29 Julian Lyndon-Smith aster...@dotr.com
 
  check out the endbeforehexten option in cdr.conf
 
  this needs to set to yes
 
  Julian
 
 
 
  Unfortunately, this doesn't help.
 
  Let's drop the hangup handler at the moment, and focus on the saving
to
  file part.
  Then my issue is I can't update CDR value is hangup exten.
 
  Here is a dialplan that illustrate this:
 
  [from-foobar]
  exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel
  ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN}
and
  ${CALLERID(num)})
same = n, Set(CDR(userfield)=foo)
same = n, Dial(SIP/foobar/${EXTEN})
same = n, Set(CDR(userfield)=bar)
same = n, Hangup()
 
  exten = h,1,Verbose(0,Entering context ${CONTEXT} from
  ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to
  ${EXTEN} and ${CALLERID(num)})
same = n,
 
ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2)
 
  My goal is to get either baz1 or baz2 value in
  /var/log/asterisk/cdr-csv/Master.csv.
 
  Typing channel originate Local/7005@from-foobar application Playback
  tt-monkeys, I can see that the line with ExecIf is run but CDR still
  contains foo value (the one set before Dial).
  The strange thing is :
  1. a CDR is written at the moment extension 7005 answers,
  2. no other CDR is added when 7005 hangs up (so can't tell how long
  extension 7005 listened to monkeys fellows).
  (Setting endbeforehexten to either yes or no has no effect on this
  behaviour.
 
 
  My question are:
  1. Is it simply possible to update CDR in hangup exten ?
  2. How can I have a CDR for the application Playback part (see above)
?
  3. Any tip or suggestion ?
 
  Cheers
 
  --
  _
  -- 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:
 

Re: [asterisk-users] Pattern matching repeating digits

2013-03-29 Thread Nathan Anderson
Eric,

Thanks; of course, this is also an option.  However, setting up a separate 
context for this type of thing with several identical Goto statements also 
strikes me as inelegant, even if it is less so. 

-- Nathan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, March 28, 2013 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pattern matching repeating digits

You are correct, it is stupid 8-)  

exten = 233,1,Goto(dial-out,${EXTEN},1)
exten = 255,1,Goto(dial-out,${EXTEN},1)

[dial-out]

exten = _XXX,1,DoStuff()
exten = _XXX,n,AndMoreStuff()
exten = _XXX,n,Dial(something)
exten = _XXX,n,Hangup

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nathan Anderson
Sent: Wednesday, March 27, 2013 2:18 AM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] Pattern matching repeating digits

'lo, all,

Is there some (possibly undocumented?) way that I can pattern-match on a 
specified number of repeating digits?  (Something similar to regular 
expressions' {})

Here's an example: let's say I have a string of things that need to be done for 
both extensions 233 and 255.  I can either...

A) Repeat the exact same code for both extensions, like so:

exten = 233,1,DoStuff()
exten = 233,n,AndMoreStuff()
exten = 233,n,Dial(something)

exten = 255,1,DoStuff()
exten = 255,n,AndMoreStuff()
exten = 255,n,Dial(something)

...which is stupid, or...

B) I can attempt code reuse for similar cases (a Good Thing[tm]), and make as 
specific of a match as possible, like so:

exten = _2[35][35],1,DoStuff()
exten = _2[35][35],n,AndMoreStuff()
exten = _2[35][35],n,Dial(something)

...but this will not only match 233 and 255, but 235 and 253 as well.

It'd be nice if there was a substitute character that meant a character that 
is exactly the same as the preceding one; for example, if R was meant to 
represent such a concept, then this would do what I want:

exten = _2[35]R,1,DoStuff()
exten = _2[35]R,n,AndMoreStuff()
exten = _2[35]R,n,Dial(something)

You could even do crazy things like chain them together (this would match 2 
and 2 and nothing else);

exten = _2[35]RRR,1,DoStuff()
exten = _2[35]RRR,n,AndMoreStuff()
exten = _2[35]RRR,n,Dial(something)

Am I missing something or does this really not exist?

Thanks,

--
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com

--
_
-- 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

--
_
-- 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] IPv6

2013-03-29 Thread Hans Witvliet
Hi all,

I had to re-install a new machine and noticed that by default, ip was
only listening on 0.0.0.0, thus ipv4 only. Easily changed.

However, when looking at iax.conf, I found here the same, but it looks
like iax is still ipv4 only?
If i change bindaddr=192.168.0.1 towards bindaddr=::, and look with
lsof -i iax is still not listening on V6.

Is iax/ipv6 still on the TODO-list ?


Hans



--
_
-- 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] IPv6

2013-03-29 Thread Matthew Jordan
On 03/29/2013 07:05 PM, Hans Witvliet wrote:
 Hi all,
 
 I had to re-install a new machine and noticed that by default, ip was
 only listening on 0.0.0.0, thus ipv4 only. Easily changed.
 
 However, when looking at iax.conf, I found here the same, but it looks
 like iax is still ipv4 only?
 If i change bindaddr=192.168.0.1 towards bindaddr=::, and look with
 lsof -i iax is still not listening on V6.
 
 Is iax/ipv6 still on the TODO-list ?
 

The IAX2 channel driver does not currently support IPv6. While it would
be a nice improvement, there are no plans to perform this work at this time.

Of course, the Asterisk open source developer community could take this
work on, which would be a welcome addition to Asterisk 12.

Matt

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



--
_
-- 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