Re: [asterisk-users] DPMA - Asterisk Realtime

[Robert Broyles]

Thanks for the update.

As an Authorized Digium Reseller - it's difficult for me to sell Digium 
phones if the customer can't use the cool features it comes with. They 
can purchase a standard IP Phone (which is what Digium phones are 
without DPMA) from other vendors for a lower price point.


So the ability to use DPMA with Asterisk RT is very important for our 
large deployments.


Anyone willing to contribute towards a bounty for this feature?

--
Robert Broyles

On 5/7/15 7:14 AM, Matthew Jordan wrote:




On Fri, May 1, 2015 at 10:43 AM, Robert Broyles 
rob...@webservicesaz.com mailto:rob...@webservicesaz.com wrote:


We love our Digium phones and DPMA - but we really need it to work
on our Realtime Platform. Otherwise we lose all the cool features
and they are just standard SIP phones.

Anyone working on a solution for this? Or anyone from Digium see
this on the roadmap?


Hey Robert -

We've had a number of requests to have the DPMA work more closely with 
Asterisk Realtime. Right now, that feature isn't planned for an 
upcoming scheduled release, but we do keep track of requests such as 
this. We've made a note of it, and we'll keep evaluating it versus 
other planned and requested features.


Thanks -

Matt

--
Matthew Jordan
Digium, Inc. | Director of Technology
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

[asterisk-users] DPMA - Asterisk Realtime

[Robert Broyles]

We love our Digium phones and DPMA - but we really need it to work on 
our Realtime Platform. Otherwise we lose all the cool features and they 
are just standard SIP phones.


Anyone working on a solution for this? Or anyone from Digium see this on 
the roadmap?



--
_
-- 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] Agent Privacy - chan_local

[Robert Broyles]

I'm trying to solve a problem I have with agents hanging up on callers 
before they even talk to them (caused by agents dropping their handset 
or something.)

What I want is something like AgentLogin() where the agent has to press 
'1' to accept the call.  Does anyone know how to get this to work with 
chan_local ?

Thanks!

Robert

-- 
_
-- 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] IP Phone recommendation

[Robert Broyles]

Sebastian Milioto wrote:
 Hi all,

 I have to install 25 IP Phone in some building. I want just basic IP 
 Phones like:


 Cisco-Linksys SPA922  u$s 146
 Grandstream GXP-2000   u$s 105
 Snom 300   u$s 119

 The most valuables parameters for me are (in importance order from 
 high to low):

 - Stability (device don't hang in any way)
 - Voice quality using G729
 - Provisioning

 So what device do you suggest according I said above?
 Is there another device which deserves attention?

 Thanks very much in advance,

 Sebastian


 
 Sebastian Milioto
 ITC
 Cid Campeadro 440
 Rio Tercero, Cordoba, Argentina
 msn: sebamili...@hotmail.com mailto:sebamili...@hotmail.com
 

We use Linksys SPA's and SNOM in our offices, and couldn't be happier.



-- 
_
-- 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] Realtime queue not work

[Robert Broyles]

Leif Neland wrote:
  

 - Original Message -
 *From:* Zhang Shukun mailto:bit...@gmail.com
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 mailto:asterisk-users@lists.digium.com
 *Sent:* Friday, January 15, 2010 11:48 AM
 *Subject:* [asterisk-users] Realtime queue not work

 hi, all

 i try to confiture realtime queue, but not work, details as below:

 Insert into queue_table(name)value('95040654321');

 INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
 '95040654321', 'SIP/1001', 2, 1);
 INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
 'SIP/1002', 2, 1);
 INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
 '95040654321', 'SIP/1003', 2, 1);

 but when i dial 95040654321 and press extension 1. error happens:

  -- Executing Queue(SIP/1003-, 950406543211)
 [Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
 to join queue '950406543211'
   == Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'

  
 No golden answers, but something to try.
  
 queue names can not be just numbers? I'd try calling the queue 
 q95040654321.
  
 Does show queues show the queue? Don't know if that's supposed to 
 work on realtime queues.
  
  
 Leif
  
Yes, from my experience -  'queue show' will show realtime queues.   
'show queues' technically is deprecated in 1.4, but should give the same 
results.

-- 
_
-- 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] Realtime queue not work

[Robert Broyles]

Zhang Shukun wrote:
 2010/1/15 Leif Neland le...@neland.dk:
   
 - Original Message -
 From: Zhang Shukun
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Sent: Friday, January 15, 2010 11:48 AM
 Subject: [asterisk-users] Realtime queue not work
 hi, all

 i try to confiture realtime queue, but not work, details as below:

 Insert into queue_table(name)value('95040654321');

 INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
 '95040654321', 'SIP/1001', 2, 1);
 INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
 'SIP/1002', 2, 1);
 INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
 '95040654321', 'SIP/1003', 2, 1);

 but when i dial 95040654321 and press extension 1. error happens:

  -- Executing Queue(SIP/1003-, 950406543211)
 [Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
 to join queue '950406543211'
   == Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'


 No golden answers, but something to try.

 queue names can not be just numbers? I'd try calling the queue
 q95040654321.
 

 Thank you for reply. i've try a95040654321 as the queue name but not
 work either.

 there was the same error in the cli.

   
 Does show queues show the queue? Don't know if that's supposed to work on
 realtime queues.


 Leif

 --
 _
 -- 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

 



   
Which version of Asterisk are you using?  Can you paste ( use 
pastebin.com please ) your extconfig.conf and  res_mysql.conf  (or 
res_odbc.conf)?
Something must not be configured right.

-- 
_
-- 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] Realtime Queue Members Not Ringing

[Robert Broyles]

Hi,

So I'm using Asterisk Realtime Queues and Queue members on 1.4.28.
I've noticed if there are no people in the queue when a call enters, 
even after a queue member enters, the call is never rang to him.

 From the debug, it seems that Asterisk is only grabbing the queue 
member list upon entering the queue. And not again until another call 
enters the queue. As a result, a caller will sit in the queue for an 
unknown amount of time until the the next caller enters the queue.

My next thought was, 'well, I'll leave the queue member in there and 
just pause him when he goes on break'... but the same thing occurs. If 
the caller enters the queue when the member is paused, Asterisk 
continues to see him paused until another call enters the queue.

So my question is this, is this fixed in 1.6.x? Or does anyone else even 
see this as a problem/bug?  Might I suggest a fix of if the queue is 
empty upon entry of a call, Asterisk checks back every 15 seconds (or 
whatever) to refresh the queue member list.  

Btw, if a 'queue show queuename' is done from CLi, or 
QUEUE_MEMBER_COUNT() is used it will grab a new member list, and the 
call will ring through. Just seems like a bad behavior.

Any thoughts?

--Robert

___
-- 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] Realtime Queue Members Not Ringing

[Robert Broyles]

Robert Broyles wrote:
 Hi,

 So I'm using Asterisk Realtime Queues and Queue members on 1.4.28.
 I've noticed if there are no people in the queue when a call enters, 
 even after a queue member enters, the call is never rang to him.

 From the debug, it seems that Asterisk is only grabbing the queue 
 member list upon entering the queue. And not again until another call 
 enters the queue. As a result, a caller will sit in the queue for an 
 unknown amount of time until the the next caller enters the queue.

 My next thought was, 'well, I'll leave the queue member in there and 
 just pause him when he goes on break'... but the same thing occurs. If 
 the caller enters the queue when the member is paused, Asterisk 
 continues to see him paused until another call enters the queue.

 So my question is this, is this fixed in 1.6.x? Or does anyone else 
 even see this as a problem/bug?  Might I suggest a fix of if the queue 
 is empty upon entry of a call, Asterisk checks back every 15 seconds 
 (or whatever) to refresh the queue member list. 
 Btw, if a 'queue show queuename' is done from CLi, or 
 QUEUE_MEMBER_COUNT() is used it will grab a new member list, and the 
 call will ring through. Just seems like a bad behavior.

 Any thoughts?

 --Robert

So 1.6.x doesn't have any improvement in this regard.

Can someone point me in the right direction here?

Maybe modify app_queue to refresh the queue member list from time to 
time? Or even after the periodic announcement would be good - anything 
would be an improvement.

Right now I have a cron running every minute to 'queue show 
queuename'  - but that's not scalable, especially after you have 
several queues you need to do this with.

What's the logic in the current system?


___
-- 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] multiple instances of asterisk on same machine

[Robert Broyles]

Or you could setup a VPS environment (perhaps openvz) and run Asterisk 
in a virtual environment.

I've done this in the past and it works well.

ram wrote:


 On Wed, Dec 30, 2009 at 5:29 PM, Saeed Akhtar 
 saeedakhtar@gmail.com mailto:saeedakhtar@gmail.com wrote:

 hi all,

 I have a little problem I'm using asterisk with opensips as
 opensips dispatches calls to asterisk. I have to use multiple
 asterisk servers but since for the time being im using 1 machine
 for testing i want to run different instances of asterisk running
 on 1 pc listening to different ports. Can someone please guide me
 how to do this? I'll be very thankful

  
 how about configuring different config files in different folders
 and run asterisk to use that config files.
  
 ram

 

 ___
 -- 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


___
-- 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] CDR_MYSQL 1.4 Database Structure

[Robert Broyles]

So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the 
database structure for cdr_mysql is:

CREATE TABLE cdr (
  calldate datetime NOT NULL default '-00-00 00:00:00',
  clid varchar(80) NOT NULL default '',
  src varchar(80) NOT NULL default '',
  dst varchar(80) NOT NULL default '',
  dcontext varchar(80) NOT NULL default '',
  channel varchar(80) NOT NULL default '',
  dstchannel varchar(80) NOT NULL default '',
  lastapp varchar(80) NOT NULL default '',
  lastdata varchar(80) NOT NULL default '',
  duration int(11) NOT NULL default '0',
  billsec int(11) NOT NULL default '0',
  disposition varchar(45) NOT NULL default '',
  amaflags int(11) NOT NULL default '0',
  accountcode varchar(20) NOT NULL default '',
  uniqueid varchar(32) NOT NULL default '',
  userfield varchar(255) NOT NULL default ''
);

Just curious if anyone has successfully patched cdr_addon_mysql to use 
accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'?
Seems logical that the cdr_mysql addon should be updated to reflect the 
current cdr. And for backwards compatibility it can still accept 
'calldate'.


Thanks in advance

___
-- 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] CDR_MYSQL 1.4 Database Structure

[Robert Broyles]

Tilghman Lesher wrote:
 On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote:
   
 So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the
 database structure for cdr_mysql is:

 CREATE TABLE cdr (
   calldate datetime NOT NULL default '-00-00 00:00:00',
   clid varchar(80) NOT NULL default '',
   src varchar(80) NOT NULL default '',
   dst varchar(80) NOT NULL default '',
   dcontext varchar(80) NOT NULL default '',
   channel varchar(80) NOT NULL default '',
   dstchannel varchar(80) NOT NULL default '',
   lastapp varchar(80) NOT NULL default '',
   lastdata varchar(80) NOT NULL default '',
   duration int(11) NOT NULL default '0',
   billsec int(11) NOT NULL default '0',
   disposition varchar(45) NOT NULL default '',
   amaflags int(11) NOT NULL default '0',
   accountcode varchar(20) NOT NULL default '',
   uniqueid varchar(32) NOT NULL default '',
   userfield varchar(255) NOT NULL default ''
 );

 Just curious if anyone has successfully patched cdr_addon_mysql to use
 accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'?
 Seems logical that the cdr_mysql addon should be updated to reflect the
 current cdr. And for backwards compatibility it can still accept
 'calldate'.
 

 The MySQL driver contains all of the same information, albeit in a slightly
 different form.  Calldate is the same as start, calldate plus duration minus
 billsec is the same as answer, and calldate plus duration is the same as end.

 Generally, we do not make design changes in the middle of a release cycle,
 especially given that such changes would break a great many existing systems.
 Given that there's no security reason why we would need to make such a change,
 it is out of the question.  While you're certainly welcome to make such a
 change on your own systems, such a change will not be committed in the 1.4
 addons.

 In the 1.6 series and forward, we've changed the mysql driver to scan the
 table metadata and adapt the queries to the table structure.  Therefore, you
 could, in fact, use 'start', 'answer', and 'end' in the 1.6 series, as you
 suggested, above, and it would work perfectly well.  On the other hand, if you
 kept the legacy structure, that would work, too.

   
Thanks for the reply.

So my next question is could I take the cdr_mysql from 1.6's addons and 
use it in 1.4?



___
-- 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] state_interface how to

[Robert Broyles]

Can anyone point me to a working how to on state_interface?

I found this little example:
http://www.freepbx.org/v2/ticket/3496

But it didn't work with the backport on v1.4.x, so I have v1.6.2 installed
on my test box, and it's still not working.

Are there any special settings that need to be set for this to work?

queues.conf:
[test]
ringinuse=no
eventmemberstatus=yes
eventwhencalled=yes
member = Local/1...@internal/n,0,,SIP/1000


sip.conf:
[1000]
type=friend
context=phones
host=dynamic
secret=cAu19lap
canreinvite=no
allow=all
nat=yes
expire=120
callcounter=5

extensions.conf:
[phones]; IP SIP Phones, allows them to call local agents.
include = internal

[internal]
exten = 1050,1,Dial(SIP/1000)


Queue always shows the agent has 'Not in Use' ...
testbox*CLI queue show
test has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime,
0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
  Local/1...@internal/n (Not in use) has taken no calls yet
   No Callers

Any ideas?
___
-- 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] state_interface backport issue

[Robert Broyles]

Any takers?
Still trying to get this resolved...

Thanks!

Robert Broyles wrote:
 It's my understanding that the backport is available now in 1.4. 
 However, seem to be having some issues with it. Just wondering if I 
 have everything setup right.

 I'm running 1.4.26.2 realtime.
 queue_members:
 `uniqueid` int(10) unsigned NOT NULL auto_increment,
  `membername` varchar(40) default NULL,
  `queue_name` varchar(128) default NULL,
  `interface` varchar(128) default NULL,
  `penalty` int(11) default NULL,
  `paused` int(1) default NULL,
  `state_interface` varchar(128) NOT NULL,

 Data:
 1, Name, QUEUENAME, Local/1...@agents/n, 1, , SIP/100

 Local agents are setup setup in an 'agents' context.

 [agents]
 exten = 1050,1,Set(agentsip=${DB(agent_sip/1050)}
 exten = 1050,2,Dial(SIP/${agentsip})

 Queue shows the agent as unavailable when the SIP device (SIP/100) is 
 down. (as I would hope)... but shows the agent as available all the 
 other times.

 As a result my CLi is on fire with 'busy' notices, because it's trying 
 to ring an agent even when they are on a call. If I remove the 
 state_interface, it shows them as 'busy' in the queue, and doesn't 
 ring them.

 Let's see, what else did I forget? Other details:

 sip.conf: limitonpeers=yes
 and call-limit=5 on each SIP device
 queue.conf: ringinuse=no

 Anything else I should look for?


 Thanks!

 -Rob



___
-- 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] state_interface backport issue

[Robert Broyles]

It's my understanding that the backport is available now in 1.4. 
However, seem to be having some issues with it. Just wondering if I have 
everything setup right.

I'm running 1.4.26.2 realtime. 

queue_members:
 `uniqueid` int(10) unsigned NOT NULL auto_increment,
  `membername` varchar(40) default NULL,
  `queue_name` varchar(128) default NULL,
  `interface` varchar(128) default NULL,
  `penalty` int(11) default NULL,
  `paused` int(1) default NULL,
  `state_interface` varchar(128) NOT NULL,

Data:
1, Name, QUEUENAME, Local/1...@agents/n, 1, , SIP/100

Local agents are setup setup in an 'agents' context.

[agents]
exten = 1050,1,Set(agentsip=${DB(agent_sip/1050)}
exten = 1050,2,Dial(SIP/${agentsip})

Queue shows the agent as unavailable when the SIP device (SIP/100) is 
down. (as I would hope)... but shows the agent as available all the 
other times.

As a result my CLi is on fire with 'busy' notices, because it's trying 
to ring an agent even when they are on a call. If I remove the 
state_interface, it shows them as 'busy' in the queue, and doesn't ring 
them.

Let's see, what else did I forget? Other details:

sip.conf: limitonpeers=yes
and call-limit=5 on each SIP device
queue.conf: ringinuse=no

Anything else I should look for?


Thanks!

-Rob

___
-- 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] Dead Call But Still Active

[Robert Broyles]

I'm having a strange issue, and not really sure where to even begin to 
troubleshoot it. First let me explain that I have all agents setup 
locally ( local/1...@agents/n)

A call will come in and ring to the agent. When the agent answers the 
call, they just hear a dial tone. Agent hangs up. Asterisk still shows 
the agent as 'in use' in queue status.  And 'show channels' shows that 
the channel is still active.

I have enabled full logging, and noticed this when it happens:

[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0
[Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples 
from write factory 0xacfbce0

What makes it more strange is that I can ChanSpy on that active channel, 
and I can hear the caller talking to themselves.

The logs also show asterisk bridging the incoming SIP connection with 
the Phone, but when the agent hangs up, it shows the agent as hanging 
up, but it's not closing the bridge.

Any ideas? Where should I begin in troubleshooting this? Thanks!

-- 
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145



DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.

___
-- 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] Overriding Queue Wrapup Time

[Robert Broyles]

So I'm guessing, I would disable any wrapup on the queue, and then in my 
'h' extension pause the agent for a set period of time, with another 
extension to unpause the agent if entered? 


Or is there a better way to set the pause after the call is over?

Thanks!

--
Regards,
Robert Broyles




Mark Michelson wrote:

Robert Broyles wrote:
  

Is there a way to override the queue wrapup time on the fly?

I would like to allow a longer wrapup time for my agents, but if they 
are already done with closing up the call ticket, I would like them to 
be able to dial an extension or something to override the wrapup.


Is there a way to do that?




There's not a way to do that using the wrapuptime of a queue member, but there 
are other ways you could potentially take care of this. For instance, you can 
pause a queue member once he has finished talking and set a timer so that the 
member will automatically become unpaused after a certain time. If the member is 
ready to receive calls again before the time has expired, he can dial an 
extension to unpause himself.


Mark Michelson

___
-- 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

  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Overriding Queue Wrapup Time

[Robert Broyles]

Is there a way to override the queue wrapup time on the fly?

I would like to allow a longer wrapup time for my agents, but if they 
are already done with closing up the call ticket, I would like them to 
be able to dial an extension or something to override the wrapup.

Is there a way to do that?

-- 
Regards,
Robert Broyles

___
-- 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] Bounty- CDR Bug Fix

[Robert Broyles]

Great backports! :-)

This should really be merged into 1.4.

--
Regards,
Robert Broyles



Atis Lezdins wrote:

Well, i can share mine backports of queue_log into mysql for 1.4.

Basically you need two backports (that's why there are numerous
files). Realtime store/destroy allows Asterisk Realtime engine to use
INSERT's on MySQL. It needs two patches - one for Asterisk, one for
Asterisk-addons (mysql part). And then there's itself queue_log
realtime patch.

I'm using it on Asterisk 1.4.19 for some half year already, so it
could be considered stable. I just tested and it does apply cleanly to
Asterisk 1.4.23 (and was previously working with latest Addons-1.4.7).

Also an advantage of this is - that it's already merged into 1.6.0, so
upgrade shouldn't be a problem.

So, some brief instructions:

1) apply 
http://ftp.iq-labs.net/realtime_store_destroy-1.4/asterisk_realtime_store_destroy_1.4.19.patch
to Asterisk
2) apply 
http://ftp.iq-labs.net/queue_log-1.4/asterisk_queue_log_realtime_1.4.19.patch
to Asterisk
make  make install

3) make dist-clean on Asterisk-addons
4) Apply 
http://ftp.iq-labs.net/realtime_store_destroy-1.4/asterisk_addons_realtime_store_destroy_1.4.6.patch
to Asterisk-addons
make  make install

* ensure that res_mysql.conf has working connection:

[general]
dbhost = localhost
dbname = asterisk
dbuser = asterisk
dbpass = pass
dbport = 3306
dbsock = /tmp/mysql.sock

* add to extconfig.conf:

queue_log = mysql,asteriskcdrdb,queue_log

* create mysql table:

CREATE TABLE queue_log (
id int(10) unsigned NOT NULL PRIMARY KEY AUTO_INCREMENT
time int(10) unsigned,
callid varchar(20),
queuename int(10) unsigned,
agent varchar(40),
event 
enum('ABANDON','ADDMEMBER','AGENTCALLBACKLOGIN','AGENTCALLBACKLOGOFF','AGENTDUMP','AGENTLOGIN','AGENTLOGOFF','COMPLETEAGENT','COMPLETECALLER','CONFIGRELOAD','CONNECT','EDITMEMBER','ENTERQUEUE','EXITEMPTY','EXITWITHKEY','EXITWITHTIMEOUT','PAUSEALL','PAUSE','QUEUESTART','REMOVEMEMBER','RINGNOANSWER','SYSCOMPAT','TRANSFER','TRANSFERATTENDED','UNPAUSE','UNPAUSEALL'),
data varchar(255)
)

Regards,
Atis

On Thu, Mar 5, 2009 at 8:54 PM, Robert Broyles rob...@poornam.com wrote:
  

The patch I was referring to is:
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queue_logging

It doesn't work for the current SVN 1.4

--
Regards,
Robert Broyles


Anthony Francis wrote:

Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a
patch way back for 1.2 that allowed all queue log events to sh,ow up in
the AMI, just haven't had time to make a new version for 1.6.

Maybe this time I can get the patch in trunk and it will always be there.

Robert Broyles wrote:


Problem is, without going to 1.6, I can't get the queue log or events
posted to MySQL in realtime.

There used to be a patch out there for queue_log, but it doesn't work
with versions 1.4.21 or higher.
--
Regards,
Robert Broyles




Anthony Francis wrote:


Robert Broyles wrote:



I saw some of the heat about the $20 bounty earlier.  So I don't want to
put a low bounty out.
Quote me a bounty, and I'll see if I can get it approved by management. :-)

I'm in need of getting this bug fixed.  Bug has all of the details, but
basically 1.4.22 broke it all.
I've waited as long as I can - hoping the bug would 'resolve itself' -
but now I'm putting a bounty out on it.

http://bugs.digium.com/view.php?id=13691





I would not recommend using CDR's for queue data, instead I use the
queue events, or at a minimum the queue log.






___
-- 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




___
-- 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






  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- Bandwidth and Colocation

Re: [asterisk-users] Bounty- CDR Bug Fix

[Robert Broyles]

Problem is, without going to 1.6, I can't get the queue log or events 
posted to MySQL in realtime.


There used to be a patch out there for queue_log, but it doesn't work 
with versions 1.4.21 or higher.


--
Regards,
Robert Broyles




Anthony Francis wrote:

Robert Broyles wrote:
  
I saw some of the heat about the $20 bounty earlier.  So I don't want to 
put a low bounty out.

Quote me a bounty, and I'll see if I can get it approved by management. :-)

I'm in need of getting this bug fixed.  Bug has all of the details, but 
basically 1.4.22 broke it all.
I've waited as long as I can - hoping the bug would 'resolve itself' - 
but now I'm putting a bounty out on it.


http://bugs.digium.com/view.php?id=13691

  

I would not recommend using CDR's for queue data, instead I use the 
queue events, or at a minimum the queue log.


  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Bounty- CDR Bug Fix

[Robert Broyles]

The patch I was referring to is:
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queue_logging

It doesn't work for the current SVN 1.4

--
Regards,
Robert Broyles



Anthony Francis wrote:
Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a 
patch way back for 1.2 that allowed all queue log events to sh,ow up in 
the AMI, just haven't had time to make a new version for 1.6.


Maybe this time I can get the patch in trunk and it will always be there.

Robert Broyles wrote:
  
Problem is, without going to 1.6, I can't get the queue log or events 
posted to MySQL in realtime.


There used to be a patch out there for queue_log, but it doesn't work 
with versions 1.4.21 or higher.

--
Regards,
Robert Broyles

  



Anthony Francis wrote:


Robert Broyles wrote:
  
  
I saw some of the heat about the $20 bounty earlier.  So I don't want to 
put a low bounty out.

Quote me a bounty, and I'll see if I can get it approved by management. :-)

I'm in need of getting this bug fixed.  Bug has all of the details, but 
basically 1.4.22 broke it all.
I've waited as long as I can - hoping the bug would 'resolve itself' - 
but now I'm putting a bounty out on it.


http://bugs.digium.com/view.php?id=13691

  


I would not recommend using CDR's for queue data, instead I use the 
queue events, or at a minimum the queue log.


  
  



___
-- 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



  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Bounty- CDR Bug Fix

[Robert Broyles]

I saw some of the heat about the $20 bounty earlier.  So I don't want to 
put a low bounty out.
Quote me a bounty, and I'll see if I can get it approved by management. :-)

I'm in need of getting this bug fixed.  Bug has all of the details, but 
basically 1.4.22 broke it all.
I've waited as long as I can - hoping the bug would 'resolve itself' - 
but now I'm putting a bounty out on it.

http://bugs.digium.com/view.php?id=13691

-- 
Regards,
Robert Broyles



DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.

___
-- 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] Bounty- CDR Bug Fix

[Robert Broyles]

Yes, I've already posted notes on the bug.
I applied the patch, and when attempting to recompile, it fails.

--
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145



Jason Parker wrote:

Robert Broyles wrote:
  
I saw some of the heat about the $20 bounty earlier.  So I don't want to 
put a low bounty out.

Quote me a bounty, and I'll see if I can get it approved by management. :-)

I'm in need of getting this bug fixed.  Bug has all of the details, but 
basically 1.4.22 broke it all.
I've waited as long as I can - hoping the bug would 'resolve itself' - 
but now I'm putting a bounty out on it.


http://bugs.digium.com/view.php?id=13691




There is already a patch on this bug that requires testing...

If you have feedback, please respond on the bug so that it can get committed for
inclusion into future releases.

If the patch works, I'm sure murf would accept a good rootbeer. :)

___
-- 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

  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Bounty- CDR Bug Fix

[Robert Broyles]

By the way, I'm more than happy to send murf a case of rootbeer (or real 
beer assuming he's legal :-P ) if this bug and/or related bugs can be 
resolved soon. :-)


--
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145



Jason Parker wrote:

Robert Broyles wrote:
  
I saw some of the heat about the $20 bounty earlier.  So I don't want to 
put a low bounty out.

Quote me a bounty, and I'll see if I can get it approved by management. :-)

I'm in need of getting this bug fixed.  Bug has all of the details, but 
basically 1.4.22 broke it all.
I've waited as long as I can - hoping the bug would 'resolve itself' - 
but now I'm putting a bounty out on it.


http://bugs.digium.com/view.php?id=13691




There is already a patch on this bug that requires testing...

If you have feedback, please respond on the bug so that it can get committed for
inclusion into future releases.

If the patch works, I'm sure murf would accept a good rootbeer. :)

___
-- 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

  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Bounty- CDR Bug Fix

[Robert Broyles]

Actually, that's alcohol abuse. :-)

Regards,
Robert Broyles



Christian Victor wrote:

2009/3/4 Atis Lezdins a...@iq-labs.net mailto:a...@iq-labs.net

Bottle of Riga Black Balsam (45%), just have to figure out a way
to send it :)


Balsam??? By mail? Doesn't that count as liquid explosive? ;-)

Chris


___
-- 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




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Bounty- CDR Bug Fix

[Robert Broyles]

Yea, that patch was tried, and doesn't resolve the issue either.
I will hold out on the bounty a little longer... maybe it will be 
resolved soon. It's pretty important for us.


--
Regards,
Robert Broyles




Jason Parker wrote:

Tilghman Lesher wrote:
  

On Wednesday 04 March 2009 10:24:16 Robert Broyles wrote:


By the way, I'm more than happy to send murf a case of rootbeer (or real
beer assuming he's legal :-P ) if this bug and/or related bugs can be
resolved soon. :-)
  

Murf is plenty legal; he simply doesn't consume alcohol.




This, of course, has nothing to do with my original point.  It was more along
the lines of no need to pay a bounty - it may already be fixed. :)

There was another patch uploaded to that bug several weeks ago that I believe
supersedes the original patch(es).  That is what I was suggesting testing.  The
comments on the bug explain it.

___
-- 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

  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] CDR - What Changed?

[Robert Broyles]

What's a reasonable bounty to put on this?
I would like the functionality that was in 1.4.19-1.4.21(?) returned.

I was holding out on the bug, waiting for a resolution, but it looks 
like the community would rather redo the way CDR works although, then 
patch a broken, flawed system. In the meantime, I need to be able to 
capture unanswered calls in the previously semi working method.


--
Regards,
Robert Broyles




Steve Murphy wrote:

On Mon, 2009-01-05 at 12:27 -0700, Robert Broyles wrote:
  

On 12/17/08 I updated to 1.4.22 from 1.4.21...

Now the CDR data isn't recording calls where the caller hung up while 
waiting on the Queue.


Sample CDR data BEFORE the upgrade:

2008-10-30 12:46:47;\John\ 
0006741103;0006741103;11621708182;incoming;SIP/carrier-3;SIP/120-09232ae0;Queue;CSR1;562;518;NO 
ANSWER;3;;1193773594.70627;


Now there's nothing in the CDR for these calls.

I dug through the ChangeLog, but didn't see anything directly related to 
this.  Any ideas?
At first I thought it was the 'unanswered' option in the cdr.conf, but 
it's set to 'yes.'


Thanks in advance for the help.



Robert--

Could this be the same as Mantis bug 13691?
(http://bugs.digium.com/view.php?id=13691)

I'm hoping to get some time and try to clear out a bunch of CDR bugs...

murf

  



___
-- 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




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Odd Read App Issues - RESOLVED

[Robert Broyles]

FYI to everyone...

It was an issue on Vitelity's end on the gateway I was assigned to. They 
switched me, and it's working fine now.


--
Regards,
Robert Broyles



Brent Davidson wrote:

Robert Broyles wrote:
I turned on DTMF debugging. It looks like the extra digits coming in 
are less than the minimum duration of 100ms


Anyone know how to force that minimum duration?

[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'1' received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'1' received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'1' received on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '1' on SIP/carrier-c4022740
[Feb 26 12:15:07] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '1' 
received on SIP/carrier-c4022740, duration 20 ms
[Feb 26 12:15:07] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '1' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'2' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'2' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'2' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '2' 
received on SIP/carrier-c4022740, duration 20 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '2' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'3' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '3' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'3' received on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '3' on SIP/carrier-c4022740
[Feb 26 12:15:08] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '3' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:08] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '3' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'4' received on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '4' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:09] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'4' received on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '4' on SIP/carrier-c4022740
[Feb 26 12:15:09] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '4' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:09] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '4' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin 
'5' received on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '5' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF

[asterisk-users] Odd Read App Issues

[Robert Broyles]

So I'm using the READ() application within an IVR, and having a strange 
issue, and wondering if anyone else has had this problem.

When calling from an outside line, and entering the digits during the 
read() part of my dialplan, it's accepting some of the digits twice, 
though it's only keyed in once.

When testing the dialplan internally, it accepts only the digits that I 
key in.

Anyone else experienced this?

-- 
Regards,
Robert Broyles




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.

___
-- 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] Odd Read App Issues

[Robert Broyles]

Btw, I'm using Asterisk SVN-branch-1.4-r178640


Robert Broyles wrote:
 So I'm using the READ() application within an IVR, and having a strange 
 issue, and wondering if anyone else has had this problem.

 When calling from an outside line, and entering the digits during the 
 read() part of my dialplan, it's accepting some of the digits twice, 
 though it's only keyed in once.

 When testing the dialplan internally, it accepts only the digits that I 
 key in.

 Anyone else experienced this?

   


DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.

___
-- 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] Odd Read App Issues

[Robert Broyles]

Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?

--
Regards,
Robert Broyles




Eric Wieling, Asteria Solutions Group wrote:

Robert Broyles wrote:
  
So I'm using the READ() application within an IVR, and having a strange 
issue, and wondering if anyone else has had this problem.


When calling from an outside line, and entering the digits during the 
read() part of my dialplan, it's accepting some of the digits twice, 
though it's only keyed in once.


When testing the dialplan internally, it accepts only the digits that I 
key in.


Anyone else experienced this?



Yes.  Most of the time it is either because I put relaxdtmf=yes in 
zapata.conf or because my rxgain is too low on that port.


  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Odd Read App Issues

[Robert Broyles]

Not at all.
In fact, I found that relaxdtmf=yes is now available for sip.conf as of 
1.4 as well.

However, that didn't resolve the problem.

--
Regards,
Robert Broyles




Eric Wieling, Asteria Solutions Group wrote:

Robert Broyles wrote:
  

Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?



Sorry for wasting your time.


  




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Odd Read App Issues

[Robert Broyles]

Yea, I tried that too. I have it: dtmfmode=rfc2833

--
Regards,
Robert Broyles


Brent Davidson wrote:

Robert Broyles wrote:

Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles

  



Eric Wieling, Asteria Solutions Group wrote:

Robert Broyles wrote:
  
So I'm using the READ() application within an IVR, and having a strange 
issue, and wondering if anyone else has had this problem.


When calling from an outside line, and entering the digits during the 
read() part of my dialplan, it's accepting some of the digits twice, 
though it's only keyed in once.


When testing the dialplan internally, it accepts only the digits that I 
key in.


Anyone else experienced this?



Yes.  Most of the time it is either because I put relaxdtmf=yes in 
zapata.conf or because my rxgain is too low on that port.



I've seen an issue similar to this when the sip peer was providing 
DTMF over multiple encodings at the same time.  Usually, it's when 
Asterisk is expecting DTMF via inband, but the peer is sending inband 
and either INFO or rfc2833.  What do you have the dtmfmode= line set 
to in your sip.conf?



___
-- 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




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] Odd Read App Issues

[Robert Broyles]

 26 12:15:10] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '5' 
received on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '5' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '5' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:10] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '5' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '6' 
received on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '6' on SIP/carrier-c4022740
[Feb 26 12:15:10] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '6' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:10] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '6' on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '6' 
received on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '6' on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '6' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:11] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '6' on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '7' 
received on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '7' on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '7' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:11] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '7' on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '7' 
received on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '7' on SIP/carrier-c4022740
[Feb 26 12:15:11] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '7' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:11] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '7' on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '8' 
received on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '8' on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '8' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:12] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '8' on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '8' 
received on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '8' on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '8' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:12] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '8' on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '8' 
received on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '8' on SIP/carrier-c4022740
[Feb 26 12:15:12] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '8' 
received on SIP/carrier-c4022740, duration 20 ms
[Feb 26 12:15:12] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '8' on SIP/carrier-c4022740
[Feb 26 12:15:13] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '9' 
received on SIP/carrier-c4022740
[Feb 26 12:15:13] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '9' on SIP/carrier-c4022740
[Feb 26 12:15:13] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '9' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:13] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '9' on SIP/carrier-c4022740
[Feb 26 12:15:13] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '9' 
received on SIP/carrier-c4022740
[Feb 26 12:15:13] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin 
ignored '9' on SIP/carrier-c4022740
[Feb 26 12:15:13] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '9' 
received on SIP/carrier-c4022740, duration 100 ms
[Feb 26 12:15:13] DTMF[3564]: channel.c:2265 __ast_read: DTMF end 
passthrough '9' on SIP/carrier-c4022740


After further testing, it seems to only be a problem when the same digit 
is entered 2 times or more in a roll.
Any of the digits received with duration of 20ms aren't supposed to be 
there, but they show up anyway.


Can someone else check this on their system, and see if this is a problem?

--
Regards,
Robert Broyles




Brent Davidson wrote:

Robert Broyles wrote:

Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles

  



Eric Wieling, Asteria Solutions Group wrote:

Robert Broyles wrote:
  
So I'm using the READ() application within an IVR

Re: [asterisk-users] trunk to trunk

[Robert Broyles]

Glad I could help!! :-D


Leonja Cerebro wrote:

To Robert Broyles,
Thank you very much, it is very helpful information.

Regards,
Leonid

2009/2/18 Robert Broyles rob...@poornam.com mailto:rob...@poornam.com

Hi,

You might want to check out this tutorial:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/

It's a good place to start.

--
Regards,
Robert Broyles






Leonja Cerebro wrote:

Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP
trunk of Asterisk B (registered in Asterisk A as extension)
to incoming call across another trunk of Asterisk B to extension
of Asterisk C
What the dial plan should be?

Thanks
-- 
We never did too much talking anyway

So don't think twice, it's all right

___
-- 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


___
-- 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




--
We never did too much talking anyway
So don't think twice, it's all right


___
-- 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




DISCLAIMER  :  This email and any files transmitted with it are property of 
Poornam Info Vision Pvt. Ltd. This email contains confidential information 
intended solely for the use of the individual or entity to whom they are 
addressed. If you have received this email in error please notify the sender 
immediately and delete this e-mail from your system. If you are not the 
intended recipient you are notified that disclosing, copying, distributing or 
taking any action in reliance on the contents of this information is strictly 
prohibited.

Warning: Although the company has taken reasonable precautions to ensure no 
viruses are present in this email, the company cannot accept responsibility for 
any loss or damage arising from the use of this email or attachments.___
-- 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] trunk to trunk

[Robert Broyles]

Hi,

You might want to check out this tutorial:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/

It's a good place to start.

--
Regards,
Robert Broyles




Leonja Cerebro wrote:

Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk 
of Asterisk B (registered in Asterisk A as extension)
to incoming call across another trunk of Asterisk B to extension of 
Asterisk C

What the dial plan should be?

Thanks
--
We never did too much talking anyway
So don't think twice, it's all right


___
-- 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
___
-- 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] AgentCallBackLogin no longer works after installing asterisk 1.6

[Robert Broyles]

Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/

Regards,
Robert Broyles

oumar ndiaye wrote:
 Hi,
  
 My queue used to work fine until I upgraded to 1.6. I am getting the 
 message:
  No application 'AgentCallBackLogin' for extension (default, 31001, 1)
 After some rearch I learnt that AgentCallBackLogin is removed in 1.6.
  
 Any one has a configuration that works in place of AgentCallBackLogin in 
 1.6.
 
 -- 
 ond
 
 
 
 
 ___
 -- 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

___
-- 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] AgentCallBackLogin no longer works after installing asterisk 1.6

[Robert Broyles]

Why don't you use followme if you want to do that?

In fact, you can have followme, plus the local agents as mentioned in 
the previous alternative that I mentioned.

--
Regards,
Robert Broyles


Anthony Francis wrote:
 Robert Broyles wrote:
 Check out this alternative:
 http://hostseries.com/agentcallbacklogin-alternative/

 Regards,
 Robert Broyles

 
 I like what he came up ,with however it doesn't replace the agent 
 callback login systems use of being able to make an agent press a key to 
 accept a call, very important when people are logging in via cell phone 
 and you don't their voice mail answering the call. In fact none of the 
 replacements do that. FAIL
 

___
-- 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] AgentCallBackLogin no longer works after installing asterisk 1.6

[Robert Broyles]

Hmm, this is all very interesting.

Looks like using a Macro and the 'M' Dial() option is the way to go for 
now if you need the answer confirmation.

http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Look at example #2, and adapt it for your needs.

--
Regards,
Robert Broyles


Philipp Kempgen wrote:
 Anthony Francis schrieb:
 Robert Broyles wrote:
 Check out this alternative:
 http://hostseries.com/agentcallbacklogin-alternative/
 
 ---cut---
 [agents]
 exten = 1050,1,Set(AGENT_SIP=${DB(agent_sip/1050)})
 exten = 1050,n,Dial(SIP/${AGENT_SIP})
 ---cut---
 
 I like what he came up ,with however it doesn't replace the agent 
 callback login systems use of being able to make an agent press a key to 
 accept a call, very important when people are logging in via cell phone 
 and you don't their voice mail answering the call. In fact none of the 
 replacements do that. FAIL
 
 First of all: The voicemail acts on behalf of the subscriber.
 If they configure their voicemail to answer the call it's their
 fault.
 
 But I understand the problem.
 http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
 talks about answer confirmation:
 ---cut---
 If the letter c follows, then Answer Confirmation is requested,
 in which the call is not considered answered until the called
 user presses #.
 ---cut---
 
 So Dial(Zap/G1c/${phone_number}) might work.
 
 Not sure why that is implemented for Zap channels only.
 It should be an option to Dial() instead.
 
 
Philipp Kempgen
 

___
-- 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] AgentCallBackLogin no longer works after installing asterisk 1.6

[Robert Broyles]

You guys... grr...
I'm still on 1.4 svn ... not ready to even think about 1.6 OR 1.8(when 
it's released) for production right now. :-)

--
Regards,
Robert Broyles


Rob Hillis wrote:
 ...except that Macros are now deprecated and will most likely be removed
 in 1.8.
 
 Robert Broyles wrote:
 Hmm, this is all very interesting.

 Looks like using a Macro and the 'M' Dial() option is the way to go for 
 now if you need the answer confirmation.

 http://www.voip-info.org/wiki-Asterisk+cmd+Dial
 Look at example #2, and adapt it for your needs.

 --
 Regards,
 Robert Broyles


 Philipp Kempgen wrote:
   
 Anthony Francis schrieb:
 
 Robert Broyles wrote:
   
 Check out this alternative:
 http://hostseries.com/agentcallbacklogin-alternative/
   
 ---cut---
 [agents]
 exten = 1050,1,Set(AGENT_SIP=${DB(agent_sip/1050)})
 exten = 1050,n,Dial(SIP/${AGENT_SIP})
 ---cut---

 
 I like what he came up ,with however it doesn't replace the agent 
 callback login systems use of being able to make an agent press a key to 
 accept a call, very important when people are logging in via cell phone 
 and you don't their voice mail answering the call. In fact none of the 
 replacements do that. FAIL
   
 First of all: The voicemail acts on behalf of the subscriber.
 If they configure their voicemail to answer the call it's their
 fault.

 But I understand the problem.
 http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
 talks about answer confirmation:
 ---cut---
 If the letter c follows, then Answer Confirmation is requested,
 in which the call is not considered answered until the called
 user presses #.
 ---cut---

 So Dial(Zap/G1c/${phone_number}) might work.

 Not sure why that is implemented for Zap channels only.
 It should be an option to Dial() instead.


Philipp Kempgen

 
 ___
 -- 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
   
 
 ___
 -- 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

___
-- 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] RFC -- Improving the quality of the mailing lists

[Robert Broyles]

I think we'd be better off posting a regular FAQ, perhaps weekly, with some of
these suggestions, as well as providing a link to that FAQ from the mailing
list signup page, along with a STRONG suggestion to peruse the FAQ first.

  

I agree with this 100%
I'm still pretty new to the mailing lists myself. I don't consider 
myself a novice Asterisk user, but one of my biggest 'complaints' is the 
lack of a well documented FAQ or Manual for Asterisk. (Unless one is 
willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - 
which quickly will be outdated again.)  I have made it a personal aim to 
document all my findings in a blog, so that it's at least searchable by 
others through Google, in hopes that others might find it useful.


But if we had a REGULARLY updated FAQ/Manual ... I think that would 
greatly cut down on the clutter posts.
___
-- 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] RFC -- Improving the quality of the mailing lists

[Robert Broyles]

I wouldn't say that voip-info.org has everything that a person would 
want to know. 
This is especially true of any recent changes to dialplan applications 
(and their available options)
Voip-info.org is a great place to start, and often you will find an 
answer there. But not always.


People are always going to ask stupid questions. There's no way to avoid 
that.  But I do believe the documentation is somewhat lacking.



Mik Cheez wrote:
It seems to me that everything one may want to know would be contained 
on voip-info.org


People don't ask stupid questions because of a lack of a FAQ to read, 
they ask stupid questions because they're too lazy do to the footwork.


Robert Broyles wrote:
  

I think we'd be better off posting a regular FAQ, perhaps weekly, with some of
these suggestions, as well as providing a link to that FAQ from the mailing
list signup page, along with a STRONG suggestion to peruse the FAQ first.

  
  

I agree with this 100%
I'm still pretty new to the mailing lists myself. I don't consider 
myself a novice Asterisk user, but one of my biggest 'complaints' is the 
lack of a well documented FAQ or Manual for Asterisk. (Unless one is 
willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - 
which quickly will be outdated again.)  I have made it a personal aim to 
document all my findings in a blog, so that it's at least searchable by 
others through Google, in hopes that others might find it useful.


But if we had a REGULARLY updated FAQ/Manual ... I think that would 
greatly cut down on the clutter posts.





___
-- 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



___
-- 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

  
___
-- 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] RFC -- Improving the quality of the mailing lists

[Robert Broyles]

Jared Smith wrote:

  On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote:
  
  
I'm still pretty new to the mailing lists myself. I don't consider
myself a novice Asterisk user, but one of my biggest 'complaints' is
the lack of a well documented FAQ or Manual for Asterisk. 

  
  
Asterisk is truly an open-source community, and that pertains to
documentation as well.  The quality and quantity of the documentation
depends heavily on contribution from the community at large.  Digium has
and will continue to put resources towards Asterisk documentation, but
every contribution from the community at large helps.

  

I understand. As someone else already mentioned, Voip-Info.org is for
more than just Asterisk. Perhaps if we created a single source that was
just for
Asterisk...where everyone could contribute towards making the
documentation better. I would be very interested in helping sponsoring
such a project, just so long as we have enough contributors. 

  
  
(Unless one is willing to buy or read O'Reilly's Book -
http://www.asteriskdocs.org - which quickly will be outdated again.)  

  
  
Alas, you've mentioned the one thing that both makes me happy and sad at
the same time.  Happy that people find it useful, and that O'Reilly was
kind enough to let us publish it under a Creative Commons license (and
put the PDF on the web for free!)... and sad that it takes so much time
and effort to keep up to date.  (And just for the record, the time that
the other authors and I spend on writing the O'Reilly book is our own
personal time -- I'm not working on it during company time!)

  

This was an excellent read. I'm sad to say that I was one that didn't
purchase the book, but made good use of the PDF. I was hoping to win
one of the books during your sessions at AstriCon this past year. Too
bad. :-( 


  
  
I have made it a personal aim to document all my findings in a blog,
so that it's at least searchable by others through Google, in hopes
that others might find it useful.

But if we had a REGULARLY updated FAQ/Manual ... I think that would
greatly cut down on the clutter posts. 

  
  
If you're interested and serious about writing, join the asterisk-docs
mailing list and let's try to get something started.  I've been beating
the documentation drum for almost seven years now, and I'd love to see
the -docs mailing list come back to life.

  

I'll be checking this out. 



___
-- 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] CDR - What Changed?

[Robert Broyles]

Anyone know how soon this will be patched?
Or are we waiting on the new CDR structure/method?

Steve Murphy wrote:

On Mon, 2009-01-05 at 12:27 -0700, Robert Broyles wrote:
  

On 12/17/08 I updated to 1.4.22 from 1.4.21...

Now the CDR data isn't recording calls where the caller hung up while 
waiting on the Queue.


Sample CDR data BEFORE the upgrade:

2008-10-30 12:46:47;\John\ 
0006741103;0006741103;11621708182;incoming;SIP/carrier-3;SIP/120-09232ae0;Queue;CSR1;562;518;NO 
ANSWER;3;;1193773594.70627;


Now there's nothing in the CDR for these calls.

I dug through the ChangeLog, but didn't see anything directly related to 
this.  Any ideas?
At first I thought it was the 'unanswered' option in the cdr.conf, but 
it's set to 'yes.'


Thanks in advance for the help.



Robert--

Could this be the same as Mantis bug 13691?
(http://bugs.digium.com/view.php?id=13691)

I'm hoping to get some time and try to clear out a bunch of CDR bugs...

murf

  



___
-- 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
___
-- 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] Snom 300 vs Grandstream gxp

[Robert Broyles]

Hi,

I've never used Snom phones, but have used the Grandstreams.
I think you will find that they just feel 'cheap.'  We had a half dozen 
of them, and the functionality is there, and they work great. But they 
just feel rough and cheap when using them.
If you are planning on using different headsets with them, you are fine. 
But if you are planning on using the factory headsets, you might find 
that the headset has rough edges, etc.

Call me 'crazy'!  We're using Linksys SPA-942/941's and couldn't be 
happier. The 941 model is a dollar or so more than the GXP, but don't 
have dual Ethernet. 942's do, for an extra $20.

Regards,
Robert Broyles


Julian Lyndon-Smith wrote:
 Can anyone who has used both comment on the pros and cons ? Need to buy 
 about 30 of these, for a small company with limited IT support.

 Julian

 __
 This email has been scanned by the MessageLabs Email Security System.
 For more information please visit http://www.messagelabs.com/email 
 __

 ___
 -- 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

   

___
-- 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] How to transfer a call from one Asterisk Server to another

[Robert Broyles]

Are you planning on connecting your two Asterisk servers with SIP or IAX?

Check out this tutorial if using SIP:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/

You should be able to adapt it to your needs. Good luck!


Paul wrote:
 Can anyone tell me how I can completely move an established call off of 
 one Asterisk server to another?
  
 In our case we have a server with our IVR.  Depending upon digits 
 entered, the call can be transferred to any of our other servers 
 depending where the extension or queue reside.
 We would like to completely move the call off of the first box so we 
 don't tie up resources on it.
  
 In our lab we are testing with 1.4.22.1
  
 Our provider which delivers inbound calls to us uses a Sonus gateway.   
 So far, testing has shown that if we transfer the inbound call prior to 
 any media playback, it works.  But, if the IVR plays media, then it is 
 failing, with a 500 internal server error being returned.
  
 Thanks for any help
  
  
  
 
 
 
 
 ___
 -- 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

___
-- 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] any SIP client for BlackBerry?

[Robert Broyles]

The Blackberry community has been begging for a SIP client for awhile. 
Apparently there are some restrictions within the Blackberry OS. But 
with the newer Blackberry models including wifi abilities, we should be 
seeing something released soon... I hope! **Fingers Crossed**


Eric Moniz wrote:
 TianLun,
 
 I should have know you would have wanted a Blackberry SIP client to 
 connect to an Asterisk box. Sorry my bad!
 I knew there was a reason why I didn't choose Truphone as my SIP client.
 I have an iPhone and I am currently using Fring which is local client 
 that connects to my Asterisk box nicely, but at this time Fring has no 
 support for the Blackberry OS.  This is why I directed you to Truphone.
 I did search the forums for a Truphone to asterisk hack, but found 
 nothing substantial.
 Keep an eye on fring.com http://fring.com maybe they will come through.
 
 Sorry best of luck!
 
 E.
 
 On Tue, Jan 6, 2009 at 2:38 PM, TianLun Song stl...@gmail.com 
 mailto:stl...@gmail.com wrote:
 
 Thank you, This one looks much better. Is it able to register with
 Asterisk instead of sign up a plan with Truphone?
 
 thank you
 
 
 On Tue, Jan 6, 2009 at 2:02 PM, Eric Moniz emoni...@gmail.com
 mailto:emoni...@gmail.com wrote:
 
 Take a look at TRUPHONE @ truphone.com http://truphone.com
  
 Eric
 
 On Tue, Jan 6, 2009 at 1:33 PM, TianLun Song stl...@gmail.com
 mailto:stl...@gmail.com wrote:
 
 Hi You all,
 
 Does anyone know any SIP client for BlackBerry?
 
 thank you
 
 -- 
 TianLun Song
 We care your day to day business operation
 CCVP, CCNP, M.Eng
 Cell:1-647-868-2950
 
 ___
 -- Bandwidth and Colocation Provided by
 http://www.api-digital.com http://www.api-digital.com/ --
 
 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 --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
 -- 
 TianLun Song
 We care your day to day business operation
 CCVP, CCNP, M.Eng
 Cell:1-647-868-2950
 
 ___
 -- 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
 
 
 
 
 
 ___
 -- 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

___
-- 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] CDR - What Changed?

[Robert Broyles]

On 12/17/08 I updated to 1.4.22 from 1.4.21...

Now the CDR data isn't recording calls where the caller hung up while 
waiting on the Queue.

Sample CDR data BEFORE the upgrade:

2008-10-30 12:46:47;\John\ 
0006741103;0006741103;11621708182;incoming;SIP/carrier-3;SIP/120-09232ae0;Queue;CSR1;562;518;NO
 
ANSWER;3;;1193773594.70627;

Now there's nothing in the CDR for these calls.

I dug through the ChangeLog, but didn't see anything directly related to 
this.  Any ideas?
At first I thought it was the 'unanswered' option in the cdr.conf, but 
it's set to 'yes.'

Thanks in advance for the help.

___
-- 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] Agents, Queues and logon/logoff

[Robert Broyles]

If you don't want to use the AEL, but want an easy way to have agents 
login and out, check out this quick tutorial:

http://hostseries.com/agentcallbacklogin-alternative/




Ariel Dorfman wrote:
 i have done some research, but there says that i can use a function called 
 AgentCallbackLogin, but it is deprecated in my system and i cant use it
 regards
 
 Ariel Dorfman a écrit :
 Hi all

 This is my first post.

 As the subject says, I need to implement on my call center the Agent
 functionality, son the agents could logon  and logoff to the queue

 How can I do this configuration? Or where can I read some info about it

 Regards

 Ariel

 To quote a forum reply i've seen today:

 It could easily be done.. Have you done any research on this, to have a
 go at it? Rather than us handing you the answer?

 You could try reading the asterisk the futur of telephony O'REILLY
 book (which is freely available), and/or the more or less offical
 asterisk wiki
http://www.voip-info.org/wiki/view/Asterisk


 ___
 -- 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
 
 ___
 -- 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

___
-- 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] CDR - What Changed?

[Robert Broyles]

Murf,

Thanks for the update. I look forward to seeing this one resolved. This 
is just the issue that I'm facing. Looks like there's a patch already 
posted on the bug. I'll wait for the bug to be closed or pushed to 
release.  Thanks again.

Robert

Steve Murphy wrote:
 On Mon, 2009-01-05 at 12:27 -0700, Robert Broyles wrote:
 On 12/17/08 I updated to 1.4.22 from 1.4.21...

 Now the CDR data isn't recording calls where the caller hung up while 
 waiting on the Queue.

 Sample CDR data BEFORE the upgrade:

 2008-10-30 12:46:47;\John\ 
 0006741103;0006741103;11621708182;incoming;SIP/carrier-3;SIP/120-09232ae0;Queue;CSR1;562;518;NO
  
 ANSWER;3;;1193773594.70627;

 Now there's nothing in the CDR for these calls.

 I dug through the ChangeLog, but didn't see anything directly related to 
 this.  Any ideas?
 At first I thought it was the 'unanswered' option in the cdr.conf, but 
 it's set to 'yes.'

 Thanks in advance for the help.
 
 Robert--
 
 Could this be the same as Mantis bug 13691?
 (http://bugs.digium.com/view.php?id=13691)
 
 I'm hoping to get some time and try to clear out a bunch of CDR bugs...
 
 murf
 
 
 
 
 
 ___
 -- 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

___
-- 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] queue log in mysql

[Robert Broyles]

With regards to storing queue_log data in mysql, it depends on the 
Asterisk service your running.

1.6.x check out the following:
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL

1.2.x  OR 1.4.x check out the following patch/solution:
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queue_logging

Hope that's what you're looking for. Good luck!


David fire wrote:
 hi
 i cant find any how to store the queue log in mysql instead of file.
 any one can send me a link or a pdf? in voip info i found how to setup a 
 realtime queue but not to store the log in mysql.
 
 end beyond the storing where i can find a good queue log parser?
 it must be opensource because is to integrate in a CRM system (vtiger) 
 wich is opensource.
  if it isnt in php there is no problem i can port it.
 
 THANKS
 David


___
-- 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] Setting Periodic-Announce filename in the dialplan

[Robert Broyles]

I'm wondering if there's a way to set which periodic-announce file is 
played from my dialplan, much like setting the monitor-filename.

Something like this:
exten = s,n, Set(PERIODIC_ANNOUNCE=foo)

This would be a great feature if it doesn't already exist. Or perhaps 
there's a better way to do this.

Thanks for your time.

-- 
Regards,
Robert Broyles

___
-- 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] Setting Periodic-Announce filename in the dialplan

[Robert Broyles]

Would this set the periodic-announce filename just for this call?
Thanks!

-- 
Regards,
Robert Broyles


___
-- 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] Setting Periodic-Announce filename in the dialplan

[Robert Broyles]

Hmm,

exten = s,1,Playback(/home/Sounds/greeting)
exten = s,n,Set(PERIODIC_ANNOUNCE=/home/Sounds/queue2)
exten = s,n,Queue(CSR)

It's not working. It just plays the default announcement.

Same goes for:
exten = s,n,Set(GLOBAL(PERIODIC_ANNOUNCE)=/home/Sounds/queue2)

Btw, I'm using v1.4.22

___
-- 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] Setting Periodic-Announce filename in the dialplan

[Robert Broyles]

That just plays back my announcement file before the caller enters the 
queue.
It's still playing the default file once in the queue.

___
-- 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] Setting Periodic-Announce filename in the dialplan

[Robert Broyles]

Thanks. The thread points to an issue with the periodic-announce not 
playing if the queue is set to ring, instead of musiconhold.
I have musiconhold with my queue.

My sample queue for testing purposes:

[CSR]
musiconhold = classic
retry = 1
strategy = ringall
joinempty = yes
periodic-announce-frequency = 15
periodic-announce = queue-periodic-announce


___
-- 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] Setting Periodic-Announce filename in the dialplan

[Robert Broyles]

Okay thank you.
This is something that I'm trying to avoid. I want to have one single 
Queue, but based on the incoming DID, have a different periodic-announce 
file played.

It would be awesome to be able to set all of the queue settings from the 
dialplan, if so wished:

examples of what I mean:
exten = s,1,Answer()
exten = s,n,Set(PERIODIC_ANNOUNCE=/home/Sounds/queue-announce)
exten = s,n,Set(PERIODIC_ANNOUNCE_FREQUENCY=60)
exten = s,n,Set(JOINEMPTY=1)
exten = s,n,Queue(CSR)

exten = 18001231234,1,Answer()
exten = 
18001231234,n,Set(PERIODIC_ANNOUNCE=/home/Sounds/toll-free-queue-announce)
exten = 18001231234,n,Set(PERIODIC_ANNOUNCE=30)
exten = 18001231234,n,Set(JOINEMPTY=1)
exten = 18001231234,n,Queue(CSR)


So that I could have all of the calls going to the same Queue, but 
depending on the DID, they are customized. Maybe this is something 
unique to my situation.

Maybe there's an easier way to do this...
Multiple queues, with ring groups, perhaps?



___
-- 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