Re: [asterisk-users] Delay in IVR

2010-06-09 Thread Sasa
Hi, sorry for my insistence but I would your aid for my problem.
Thanks.

--

   Salvatore.


- Original Message - 
From: Sasa s...@shoponweb.it
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, June 03, 2010 9:51 AM
Subject: Re: [asterisk-users] Delay in IVR


 Hi, in trixbox I don't know what create an extension with letter but only
 with number.
 Thanks.

 --

   Salvatore.



 - Original Message - 
 From: Kingsley Tart kings...@skymarket.co.uk
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, June 02, 2010 5:34 PM
 Subject: Re: [asterisk-users] Delay in IVR


 On Mon, 2010-05-24 at 14:41 +0100, Kingsley Tart wrote:
 On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote:
  HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination
  call
  is always a ring group called '600', my problem is that after press 1
  (but
  this problem is present also with press 2) before that the inbound 
  call
  is
  transfer to extension pass 10/11 seconds !
  In attach log file about incoming call.
  I use Trixbox with Asterisk-1.6.0.10.

 I know nothing of Trixbox but I had a problem with my own dialplan where
 there was a delay with the user selecting 0 from my IVR menu. It turned
 out that because my extensions all started with 0 (they were real phone
 numbers), asterisk thought that the caller might be starting to type one
 of the valid extensions and so waited for the timeout (digit timeout I
 think) before it went further.

 To see if that's your problem, try seeing whether a menu option that
 won't match the start of any of your defined extensions happens more
 quickly.

 I got around it by having the IVR in a different context where the
 extensions started with a letter, so no entered digits would match.

 Salvatore,

 Did this help?

 -- 
 Cheers,
 Kingsley.


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[asterisk-users] Out of Office

2010-06-09 Thread doug
I will be out of the office starting
Wed June 9th and returning Wed June 16th.
Please contact Mary at m...@accessgate.net cell 407-267-1463
or Jonathan at jsny...@accessgate.net cell 407-267-0056
or call our main number 888-227-9337.




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[asterisk-users] AMI Queue information about incoming call's channel before link

2010-06-09 Thread Alexandr Krylovskiy
Hi,
I'm developing an application using AMI and I need to get information
about incoming call _before_ queue member answers it.
I'm using static members (queue is pretty simple) and don't want to use
chan_agent (I think AgentCalled event will do what I'm looking for).

Here is what I'm getting now:
1. Newchannel event for incoming call followed by Newstate and Join. All these
events can be identified with uniqueid 001
2. Newchannel event for outgoing channel to queue member followed by Newstate
State: Ringing = that's what I need. But this channel (of course) has another
uniquied 002.
3. Link. That's the place where channels are get bridged and I can find out
which member (by channel) answered the call (002 answered 001 in this example).

The problem is that I need to get information about call 001 at stage 2 that
is currently possible only at stage 3, when channels get bridged.
Any ideas?

-- 
Alexandr Krylovskiy

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Re: [asterisk-users] Out of Office

2010-06-09 Thread Zeeshan Zakaria
How annoying this is.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-09 3:59 AM, d...@accessgate.net wrote:

I will be out of the office starting
Wed June 9th and returning Wed June 16th.
Please contact Mary at m...@accessgate.net cell 407-267-1463
or Jonathan at jsny...@accessgate.net cell 407-267-0056
or call our main number 888-227-9337.




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[asterisk-users] get Asterisk version from within dialplan

2010-06-09 Thread Vieri
Simple enough:
How can I get Asterisk version from within my dialplan? (preferably without 
calling an AGI script that parses asterisk -rx show version)
Is it available as a global variable?

Vieri



  

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[asterisk-users] PSTN-IVR call

2010-06-09 Thread nikhil singhania
hi all,
 I am calling a PSTN and trying to transfer it to another asterisk server
through exec_dial function.
   $agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL);
Though this is the function written by me in  a file inbound.php which is
called when an extension is dialled.
When ever i dial through PSTN it gives beeps sound, but without this line
program runs smoothly.
  Can someone help???

-- 
Nikhil Kumar
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
 niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/




-- 
Nikhil Kumar
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
 niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/
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Re: [asterisk-users] Out of Office

2010-06-09 Thread Danny Nicholas
Let's all send John and Mary an email to tell them how thoughtful Doug is
and you can bet he will either turn off or modify his rule :-)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Wednesday, June 09, 2010 4:37 AM
To: d...@accessgate.net; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Out of Office

 

How annoying this is.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-09 3:59 AM, d...@accessgate.net wrote:

I will be out of the office starting
Wed June 9th and returning Wed June 16th.
Please contact Mary at m...@accessgate.net cell 407-267-1463
or Jonathan at jsny...@accessgate.net cell 407-267-0056
or call our main number 888-227-9337.




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Re: [asterisk-users] PSTN-IVR call

2010-06-09 Thread Danny Nicholas
Since you have some inherent timeouts due to technology jumps, you might try

$agi-exec_dial(SIP,ww2001:j0...@172.26.48.62:5060,NULL,NULL,NULL);

To make asterisk wait 1 second before trying the call.

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nikhil
singhania
Sent: Wednesday, June 09, 2010 7:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PSTN-IVR call

 

 

hi all,
 I am calling a PSTN and trying to transfer it to another asterisk server
through exec_dial function.
   $agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL);
Though this is the function written by me in  a file inbound.php which is
called when an extension is dialled.
When ever i dial through PSTN it gives beeps sound, but without this line
program runs smoothly. 
  Can someone help???

-- 
Nikhil Kumar
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
 niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/




-- 
Nikhil Kumar
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
 niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/

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[asterisk-users] OT - Astmanproxy download broken ?

2010-06-09 Thread Olivier
Hi,

Is Astmanproxy still downloadable ?
At the moment, I can't download anything.
I'm usually using this http://github.com/davetroy/astmanproxy/tarball/masterURL

I can use a previous tar file but I would be pleased to know if I should do
something around this issue or not.

Regards
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Re: [asterisk-users] get Asterisk version from within dialplan

2010-06-09 Thread Jim Dickenson
Starting with version 1.6.x there is a VERSION function that I think will give 
you the version number.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jun 9, 2010, at 5:19 AM, Vieri wrote:

 Simple enough:
 How can I get Asterisk version from within my dialplan? (preferably without 
 calling an AGI script that parses asterisk -rx show version)
 Is it available as a global variable?
 
 Vieri
 
 
 
 
 
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Re: [asterisk-users] Delay in IVR

2010-06-09 Thread mike mosier
I use to use trixbox its basically asterisk with free pbx. What are your
extension numbers? Ring group number? What processor are you using? The more
info the better. When I used trixbox I never had this problem. It could be
DTMF, what is your dtmf in the trunk. What kind of trunk? Sip? What kind of
phones. What is the drtmf setting on your phones? What kind of phone are you
testing this with? I always have one test sip trunk that I know works great
for testing.

Its not likely in the code of the ivr. The problem is in how you setup
everything else that leads to trunk.

Respectfully
Michael D Mosier
Ftoc Certified

On Jun 9, 2010 2:53 AM, Sasa s...@shoponweb.it wrote:

Hi, sorry for my insistence but I would your aid for my problem.
Thanks.

--

  Salvatore.



- Original Message -
From: Sasa s...@shoponweb.it
To: Asterisk Users Mailing List - ...

Sent: Thursday, June 03, 2010 9:51 AM
Subject: Re: [asterisk-users] Delay in IVR


 Hi, in trixbox ...
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[asterisk-users] SIP Witch

2010-06-09 Thread Matthew J. Roth
Is anyone out there using SIP Witch in conjunction with Asterisk?  It claims to 
be able to enhance existing IP-PBX solutions such as Asterisk, so maybe it 
can be used as a simple means to provide secure/encrypted calls.

GNU SIP Witch - Summary http://savannah.gnu.org/projects/sipwitch
GNU SIP Witch - GNU Telephony 
http://www.gnutelephony.org/index.php/GNU_SIP_Witch
Features/SIP Witch Domain Telephony 
http://fedoraproject.org/wiki/Features/SIP_Witch_Domain_Telephony
Secure VoIP, GNU SIP Witch, and replacing Skype with free software
http://www.linuxtoday.com/it_management/2009082702235OSNT

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?

2010-06-09 Thread Jonas Kellens

I have commented out case 5, case 2 and case 3, leaving case 1, 4,6,7,8,9.

But when I press 1 on the menu, I hear:  I'm sorry, I did not 
understand your response



if (play_auto) {
cmd = '1';
} else {
cmd = vm_intro(chan, vmu, vms);
}

vms.repeats = 0;
vms.starting = 1;
while ((cmd  -1)  (cmd != 't')  (cmd != '#')) {
/* Run main menu */
switch (cmd) {
case '1':
vms.curmsg = 0;
/* Fall through */
/* commented out from here
case '5':
cmd = vm_browse_messages(chan, vms, vmu);
break;

snip code

if (vms.repeats  3)
cmd = 't';
}
}
if (cmd == 't') {
cmd = 0;
vms.repeats = 0;
}
break;
commented out till here */


Jonas.



On 06/07/2010 04:22 PM, Glenn O Larsen wrote:

On Mon, Jun 7, 2010 at 2:15 PM, Jonas Kellensjonas.kell...@telenet.be  wrote:
   

I made some changes to app_voicemail.c and recompiled asterisk. Now my
caller is only presented with the menu-choice I want.

However, the caller can still give another dtmf-input and be taken to that
specific menu.

How can I disable dtmf-input 2,3,4 if I only want the menu behind option 1
available ?
 

In the C code, find:

 case '2': /* Change folders */
 if (useadsi)
 adsi_folders(chan, 0, Change to folder...);
 cmd = get_folder2(chan, vm-changeto, 0);

then add the lines:

+   cmd = 0; /* Go back to root menu */
+   break; /* don't continue */

so it look like this:

 case '2': /* Change folders */
 cmd = 0; /* Go back to root menu */
 break; /* don't continue */
 if (useadsi)
 adsi_folders(chan, 0, Change to folder...);
 cmd = get_folder2(chan, vm-changeto, 0);


add the same lines to case 3 and 4..

Did that help?

   
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Re: [asterisk-users] Delay in IVR

2010-06-09 Thread Sasa
Hi, here information request:

extension number is 100/101
ring group number is 600
cpu : Intel(R) Pentium(R) D CPU 3.00GHz 3 GHz

On another voip machine (always with Trixbox) I haven't this problem, I have 
tried with another phones and with XLite I have always this problem.

About DTMF in SIP trunk I have (in USER Details) this parameter:
dtmfmode=rfc2833

Another sip trunk configuration is:

PEER Details:
secret=yqyqyq
nat=no
context=from-pstn
host=x.x.x.x
insecure=very
type=friend
username=yyy

In USER Details:
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
insecure=very
nat=no
port=5060
type=user

I hope that informtions are enough for resolve my problem
Thanks.

--

   Salvatore.



- Original Message - 
From: mike mosier trixbo...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, June 09, 2010 3:36 PM
Subject: Re: [asterisk-users] Delay in IVR


I use to use trixbox its basically asterisk with free pbx. What are your
 extension numbers? Ring group number? What processor are you using? The 
 more
 info the better. When I used trixbox I never had this problem. It could be
 DTMF, what is your dtmf in the trunk. What kind of trunk? Sip? What kind 
 of
 phones. What is the drtmf setting on your phones? What kind of phone are 
 you
 testing this with? I always have one test sip trunk that I know works 
 great
 for testing.

 Its not likely in the code of the ivr. The problem is in how you setup
 everything else that leads to trunk.

 Respectfully
 Michael D Mosier
 Ftoc Certified

 On Jun 9, 2010 2:53 AM, Sasa s...@shoponweb.it wrote:

 Hi, sorry for my insistence but I would your aid for my problem.
 Thanks.

 --

  Salvatore.



 - Original Message -
 From: Sasa s...@shoponweb.it
 To: Asterisk Users Mailing List - ...

 Sent: Thursday, June 03, 2010 9:51 AM
 Subject: Re: [asterisk-users] Delay in IVR


 Hi, in trixbox ...




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Re: [asterisk-users] PSTN-IVR call

2010-06-09 Thread Steve Edwards

On Wed, 9 Jun 2010, nikhil singhania wrote:


 I am calling a PSTN and trying to transfer it to another asterisk server 
through exec_dial function.
   $agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL);
Though this is the function written by me in  a file inbound.php which is 
called when an extension is dialled.


If you enable AGI debugging, this may give you a clue. Posting the console 
output may help someone help you.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000-- 
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Re: [asterisk-users] SIP Witch

2010-06-09 Thread Michelle Dupuis
I checked out the sites and can't figure out what this thing is!  (Without 
delving into the documentation).

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew J. Roth 
[mr...@imminc.com]
Sent: Wednesday, June 09, 2010 9:49 AM
To: Asterisk Users List
Subject: [asterisk-users] SIP Witch

Is anyone out there using SIP Witch in conjunction with Asterisk?  It claims to 
be able to enhance existing IP-PBX solutions such as Asterisk, so maybe it 
can be used as a simple means to provide secure/encrypted calls.

GNU SIP Witch - Summary http://savannah.gnu.org/projects/sipwitch
GNU SIP Witch - GNU Telephony 
http://www.gnutelephony.org/index.php/GNU_SIP_Witch
Features/SIP Witch Domain Telephony 
http://fedoraproject.org/wiki/Features/SIP_Witch_Domain_Telephony
Secure VoIP, GNU SIP Witch, and replacing Skype with free software
http://www.linuxtoday.com/it_management/2009082702235OSNT

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Limit total length of calls to a specifig SIP peer

2010-06-09 Thread Laurent CARON
On 08/06/2010 19:19, Steve Edwards wrote:
 The ONLY way (how's that for humble) to do this in a reliable and robust
 method is to use a real database. Personally, I like MySQL and I prefer to
 do database work in an AGI in a compiled language like C.

 Maintaining the accumulated duration in a global variable will fail if you
 need to restart Asterisk at any time. A global variable will also fail
 if you have more than 1 call finish at the same time.

 Parsing log files is guaranteed to be a resource pig and still has race
 conditions.


Hi,

I'm gonna follow your advice and store the CDR in a PostgreSQL database. 
It will allow to easily plug an AGI script to it.

Thanks

Laurent

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Re: [asterisk-users] Out of Office

2010-06-09 Thread Zeeshan Zakaria
Good idea. I am emailing them right now.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-09 9:11 AM, Danny Nicholas da...@debsinc.com wrote:

 Let’s all send John and Mary an email to tell them how thoughtful Doug is
and you can bet he will either turn off or modify his rule J


 --

*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent:* Wednesday, June 09, 2010 4:37 AM
*To:* d...@accessgate.net; Asterisk Users Mailing List - Non-Commercial
Discussion
*Subject:* Re: [asterisk-users] Out of Office





How annoying this is.

Zeeshan A Zakaria

--
www.ilovetovoip.com

 On 2010-06-09 3:59 AM, do...

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[asterisk-users] 1.6 how to use groupcount and counteronpeer in queues to avoid ringinuse

2010-06-09 Thread nik600
Dear all

i'm planning an upgrade of some asterisk installation from 1.4.32 to
1.6.0.28 (as i think it should be the most stable now).

Reading the UPGRADE-1.6.txt file i've noticed that:

* SIP: The call-limit option is marked as deprecated. It still works
in this version of
  Asterisk, but will be removed in the following version. Please use
the groupcount functions
  in the dialplan to enforce call limits. The limitonpeer
configuration option is
  now renamed to counteronpeer.

As i've experienced some problem with 1.4 release about call-limit,
i'd like to test this new counteronpeer functionality, but how to
handle the ringinuse parmeter in queues.conf ?

Basically i need that a sip user can make and receive more than one
call (like a call-limit 3 setting) but i don't want that this
interface rings if it is in a queue.

Is it possible to do that? How?

Thanks to all

-- 
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nik600
http://www.kumbe.it

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Re: [asterisk-users] own Caller ID

2010-06-09 Thread Edwin Quijada

Just is PRI line you can do it..

*---* 
*-Edwin Quijada 
*-Developer DataBase 
*-JQ Microsistemas 
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*




 
 Date: Tue, 8 Jun 2010 12:44:07 -0700
 From: asterisk@sedwards.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] own Caller ID
 
 On Tue, 8 Jun 2010, taimur hasan wrote:
 
  I want to use my own caller id, instead of the caller id of PSTN line,  
  for the outbound calls through DAHDI channel. Is there any way ??
 
 It depends on your technology (POTS, PRI, etc) and your provider.
 
 Tell your provider you want to set the outgoing caller ID and see what 
 their response is.
 
 -- 
 Thanks in advance,
 -
 Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
 Newline Fax: +1-760-731-3000
  
_

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[asterisk-users] [compat] section in asterisk.conf : compatibility with pipe delimiter

2010-06-09 Thread nik600
Dear all

after an upgrade to 1.6 from 1.4 (as explained in the UPGRADE-1.6.txt
file) the | delimiter is not working by default.

I've added a compat section in asterisk.conf a

[options]
dontwarn = yes

[compat]
pbx_realtime=1.4
res_agi=1.4
app_set=1.4

And restarted Asterisk, but i still have problem to have the |
delimiter working,

[Jun  9 23:20:54] DEBUG[11744]: pbx.c:3122 pbx_extension_helper:
Launching 'Queue'
-- Executing [...@queues:4] Queue(SIP/PL1999-0003,
queue_130) in new stack
[Jun  9 23:20:54] DEBUG[11744]: app_queue.c:4804 queue_exec: NO
QUEUE_PRIO variable found. Using default.
[Jun  9 23:20:54] DEBUG[11744]: app_queue.c:4841 queue_exec: queue:
queue_130, options: (null), url: (null), announce: (null),
expires: 0, priority: 0
[Jun  9 23:20:54] WARNING[11744]: app_queue.c:4853 queue_exec: Unable
to join queue 'queue_130'

It seems that Asterisk ignores the | delimiter, if i try with the
comma it works.

Reading the the upgrade file it seems that the pbx_realtime should
affect also the extension.conf settings... where am i wrong?

Thanks to all in advance

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nik600
http://www.kumbe.it

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Re: [asterisk-users] SIP Witch

2010-06-09 Thread Matthew J. Roth
 Matthew J. Roth wrote:
 
 Is anyone out there using SIP Witch in conjunction with Asterisk? It
 claims to be able to enhance existing IP-PBX solutions such as
 Asterisk, so maybe it can be used as a simple means to provide
 secure/encrypted calls.
 
 GNU SIP Witch - Summary http://savannah.gnu.org/projects/sipwitch
 GNU SIP Witch - GNU Telephony
 http://www.gnutelephony.org/index.php/GNU_SIP_Witch Features/SIP Witch
 Domain Telephony
 http://fedoraproject.org/wiki/Features/SIP_Witch_Domain_Telephony
 Secure VoIP, GNU SIP Witch, and replacing Skype with free software
 http://www.linuxtoday.com/it_management/2009082702235OSNT
 
 
 Michelle Dupuis wrote:
 
 I checked out the sites and can't figure out what this thing is!
 (Without delving into the documentation).

Michelle,

It was a bit unclear to me, as well.  That's why I was curious if anyone was 
actually using it.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?

2010-06-09 Thread C. Chad Wallace

At 4:04 PM on 09 Jun 2010, Jonas Kellens wrote:

 I have commented out case 5, case 2 and case 3, leaving case 1,
 4,6,7,8,9.
 
 But when I press 1 on the menu, I hear:  I'm sorry, I did not 
 understand your response

Looks like someone broke the first rule of Optimization Club[1].  I
think you need to copy these two lines from case 5 into case 1:

  cmd = vm_browse_messages(chan, vms, vmu);
  break;

It just so happens that cases 1 and 5 run the same command, so whoever
wrote it took advantage of that, optimizing the size of the binary
while reducing maintainability.


[1]
http://perlbuzz.com/mechanix/2008/02/the-rules-of-optimization-club.html



  if (play_auto) {
  cmd = '1';
  } else {
  cmd = vm_intro(chan, vmu, vms);
  }
 
  vms.repeats = 0;
  vms.starting = 1;
  while ((cmd  -1)  (cmd != 't')  (cmd != '#')) {
  /* Run main menu */
  switch (cmd) {
  case '1':
  vms.curmsg = 0;
  /* Fall through */
 /* commented out from here
  case '5':
  cmd = vm_browse_messages(chan, vms, vmu);
  break;
 
 snip code
 
  if (vms.repeats  3)
  cmd = 't';
  }
  }
  if (cmd == 't') {
  cmd = 0;
  vms.repeats = 0;
  }
  break;
 commented out till here */


-- 

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



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[asterisk-users] Out of Office

2010-06-09 Thread doug
I will be out of the office starting
Wed June 9th and returning Wed June 16th.
Please contact Mary at m...@accessgate.net cell 407-267-1463
or Jonathan at jsny...@accessgate.net cell 407-267-0056
or call our main number 888-227-9337.




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[asterisk-users] CDR in case of CallForwarding

2010-06-09 Thread srinivas Antarvedi
Hello users,

i am looking for a solution in terms of CDR for the outbound only call.

presently i have the following setup.

//extensions.conf

[from-outside]

exten = _X.,1,NoOp(IncomingCall)
exten = _X.,n,BackGround(choce.wav)
exten = _X.,n,WaitExten(5)
exten = _X.,n,Hangup

exten = _1XX.,n,NoOp(1XX series Dialing)
exten = _1XX.,n,Dial(SIP/${EXTEN},60,rg)
exten = _1XX.,n,NoOp(${DIALSTATUS})
exten = _1XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} = CONGESTION
|  ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ] ?dialmobile:end)
exten  = _1XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE mobileNUMBER
FOR THE us...@ougoingprovider,60,r)
exten = _1XX.,n(end),Hangup()


exten = _2XX.,n,NoOp(2XX series Dialing)
exten = _2XX.,n,Dial(SIP/${EXTEN},60,rg)
exten = _2XX.,n,NoOp(${DIALSTATUS})
exten = _2XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} = CONGESTION
|  ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ] ?dialmobile:end)
exten  = _2XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE mobileNUMBER
FOR THE us...@ougoingprovider,60,r)
exten = _2XX.,n(end),Hangup()


//sip.conf
[outgoingprovider]
username=X
secret=y
port=
host=dfdfddf
fromuser=


- i am planning to take  the number of calls made and the minutes spent
incase of mobile call forwarding
   as it uses my outbound trunk by giving the accountcode set to a
particular call.

- but i am getting the total call (sip call + mobile call) as a single
record in my cdr record for a given accountcode.

- i need to get something like SIP/mobilenumber either in lastdata or
dstchannel  associated accountcode as a separate cdr entry.
   i tried with disabling cdr using NoCDR for the SIP call but for the
mobile call if i use ResetCDR()  also i am totally
   losing the callrecord.

- i tried with the ForkCDR() too but of no use..

is my requirement can be fulfilled by tweaking some changes in the
extensions.conf functions/applications??

please advise as i need to bill the user for the outbound calls only...

any help is sincerely appreciated. thanks in advance.

srinivas
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Re: [asterisk-users] CDR in case of CallForwarding

2010-06-09 Thread Vardan Harutyunyan
Hello
I have also became like this problems and have found solution to make 
outgoing calls via local channel, and now if my customer do a transfer, 
I can calculate extra international outgoing calls.



-- 
Vardan Harutyunyan,
Senior System Administrator

Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com

srinivas Antarvedi wrote:
 Hello users,

 i am looking for a solution in terms of CDR for the outbound only call.

 presently i have the following setup.

 //extensions.conf

 [from-outside]

 exten = _X.,1,NoOp(IncomingCall)
 exten = _X.,n,BackGround(choce.wav)
 exten = _X.,n,WaitExten(5)
 exten = _X.,n,Hangup

 exten = _1XX.,n,NoOp(1XX series Dialing)
 exten = _1XX.,n,Dial(SIP/${EXTEN},60,rg)
 exten = _1XX.,n,NoOp(${DIALSTATUS})
 exten = _1XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} =
 CONGESTION |  ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ]
 ?dialmobile:end)
 exten  = _1XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE
 mobileNUMBER FOR THE us...@ougoingprovider,60,r)
 exten = _1XX.,n(end),Hangup()


 exten = _2XX.,n,NoOp(2XX series Dialing)
 exten = _2XX.,n,Dial(SIP/${EXTEN},60,rg)
 exten = _2XX.,n,NoOp(${DIALSTATUS})
 exten = _2XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} =
 CONGESTION |  ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ]
 ?dialmobile:end)
 exten  = _2XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE
 mobileNUMBER FOR THE us...@ougoingprovider,60,r)
 exten = _2XX.,n(end),Hangup()


 //sip.conf
 [outgoingprovider]
 username=X
 secret=y
 port=
 host=dfdfddf
 fromuser=


 - i am planning to take  the number of calls made and the minutes spent
 incase of mobile call forwarding
 as it uses my outbound trunk by giving the accountcode set to a
 particular call.

 - but i am getting the total call (sip call + mobile call) as a single
 record in my cdr record for a given accountcode.

 - i need to get something like SIP/mobilenumber either in lastdata or
 dstchannel  associated accountcode as a separate cdr entry.
 i tried with disabling cdr using NoCDR for the SIP call but for the
 mobile call if i use ResetCDR()  also i am totally
 losing the callrecord.

 - i tried with the ForkCDR() too but of no use..

 is my requirement can be fulfilled by tweaking some changes in the
 extensions.conf functions/applications??

 please advise as i need to bill the user for the outbound calls only...

 any help is sincerely appreciated. thanks in advance.

 srinivas



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