Re: [asterisk-users] Queue Breakout Input being Ignored

2011-08-14 Thread Mike Beirne 
On 8/14/2011 6:09 PM, Anton Panetta wrote:
> Hello,
> 
> Raw stats:
> Version:1.8.3.2
> OS:Centos 5.6
> Special setup: postgre database
> 
> 
> I am having a few queue issues with Asterisk specifically relating to
> breaking out from queues while on hold.
> 
> The intent is that while someone is on hold they can press a key (lets
> say *) to break from the queue and go elsewhere (in this case to leave a
> message).
> 
> However In all of my testing I am unable to get this to work, or even to
> be able to leave the queue while on hold (to go to another queue for
> example, or force any action, eg hangup, play message, etc).
> 
> I have enabled DTMF debug and I can see in the console that asterisk is
> receiving the correct information, it just appears to be ignoring it.
> 
> Tested this on 1.6 as well and got the same results ( I had assumed it
> was a bug in 1.6) everything I look at suggests what I am doing -should-
> work.
> 
> I suspect I am missing some over arching setting or something incredibly
> simple but for the life of me I cannot get this function to work as
> described.
> 
> Here is some of the  config which I beleive is relevent.
> 
> Various variants have been tested with different actions to occur (sorry
> I dont have a simpler version of the below at hand)
> 
> [extensions.conf]
> 
> [app-helpdesk-bh]
> exten => s,1(unanswered),Ringingsame  =>   n,Wait(2)
> same  =>   n,Answer
> same  =>   n,Wait(1)
> same  =>   n(answered),NoOp(Helpdesk)
> same  =>   n,Gosub(app-filteranon,s,1)
> same  =>   n,Set(CALLERID(name)=${client} Helpdesk)
> same  =>   n,NoOp(Caller ID set to: ${CALLERID(name)})
> same  =>   n,NoOp(Callers waiting in queue:
> ${QUEUE_WAITING_COUNT(helpdesk)})
> ; play the announcement for this helpdesk client (or the general intro)
> same  =>   n(announceandwait),Gosub(app-helpdesk-${client}-intro,s,1)
> same  =>   n,Gosub(app-helpdesk-${client}-special-rva,s,1)
> ; queue the call
> ;   t = allow the called party to transfer the call
> ;   k = allow the called party to park the call
> ;   c = continue on after call (to update devstate)
> same  =>   n,Set(DEVICE_STATE(Custom:helpdesk)=RINGING)
> same  =>   n,Queue(helpdesk,tk,,,600,,,app-update-helpdesk-queue-devstate)
> same  =>   n,Gosub(app-update-helpdesk-queue-devstate,s,1)
> ; if the caller can't get into the queue or is kicked out of it, go to
> ; the after hours answering service
> same  =>   n,Goto(app-helpdesk-hutchison,s,answered)
> same  =>   n(done),NoOp
> 
> ; update devstate after a hangup
> exten => h,1,Gosub(app-update-helpdesk-queue-devstate,s,1)
> 
> include => app-helpdesk-breakout
> 
> [app-helpdesk-breakout]
> exten => 1,1,Gosub(app-helpdesk-callback,s,1)
> same  =>   n,Hangup
> 
> [app-helpdesk-callback]
> ; Callers brought to this context will be able to leave a message
> ; which will then be attached to an email and sent to the appropriate
> ; helpdesk RT queue
> exten => s,1,NoOp(Helpdesk callback not implemented yet)
> same  =>   n,Return
> same  =>   n,GotoIf($["${CALLERID(num)}" = "anonymous" or
> ${CALLERID(num)} = "0"]?collect)
> 
> same  =>   n(collect),Goto(app-helpdesk-callback-collect,s,1)
> 

If I remember correctly, it only checks for key presses at the end of
each retry period. The "retry" parameter is set in queues.conf and
should be less than the timeout parameter. You have the timeout for the
queue call set to "600" or ten minutes. What is the "retry" parameter in
queues.conf set to?

Mike

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Re: [asterisk-users] DAHDI-linux-complete on CENTOS kernel 2.6.32

2011-08-14 Thread Patrick Lists 

On 08/15/2011 06:04 AM, Shaun Ruffell wrote:

On Mon, Aug 15, 2011 at 05:21:36AM +0200, Patrick Lists wrote:

On 08/14/2011 10:31 PM, Shaun Ruffell wrote:
[snip]

While I can't say I've run against that particular CENTOS kernel version,
I would be very surprised (and interested to know) if you had any
problems.


Iirc that kernel (2.6.32-71.el6.i686) is a stock Red Hat Enterprise
Linux 6 (or CentOS 6) kernel. Not sure if I understand you. Are you
saying that Digium does not test their software on RHEL6/CentOS6 or
that you tested it on RHEL6/CentOS6 but just not against the stock
kernel?


I was only saying that I personally have not tested it against that
kernel version (RHEL6 / CentOS6). My personal CentOS test machines are
running CentOS 4.9, 5.5, and 5.6. I don't yet have a 6.0 version in the
mix.

However, given that I've run on a 2.6.32 kernel (debian and vanilla) and
I would be very surprised if you have any problems compiling and running
from the source repositories.

If you're worried about it, you could wait for CentOS 6.0 packages to
show up on http://packages.asterisk.org/centos but I'm not sure what the
expected time frame for that is.


Thank you for your explanation. All clear now.

Regards,
Patrick



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Re: [asterisk-users] DAHDI-linux-complete on CENTOS kernel 2.6.32

2011-08-14 Thread Shaun Ruffell 
On Mon, Aug 15, 2011 at 05:21:36AM +0200, Patrick Lists wrote:
> On 08/14/2011 10:31 PM, Shaun Ruffell wrote:
> [snip]
> > While I can't say I've run against that particular CENTOS kernel version,
> > I would be very surprised (and interested to know) if you had any
> > problems.
> 
> Iirc that kernel (2.6.32-71.el6.i686) is a stock Red Hat Enterprise
> Linux 6 (or CentOS 6) kernel. Not sure if I understand you. Are you
> saying that Digium does not test their software on RHEL6/CentOS6 or
> that you tested it on RHEL6/CentOS6 but just not against the stock
> kernel?

I was only saying that I personally have not tested it against that
kernel version (RHEL6 / CentOS6). My personal CentOS test machines are
running CentOS 4.9, 5.5, and 5.6. I don't yet have a 6.0 version in the
mix.

However, given that I've run on a 2.6.32 kernel (debian and vanilla) and
I would be very surprised if you have any problems compiling and running
from the source repositories.

If you're worried about it, you could wait for CentOS 6.0 packages to
show up on http://packages.asterisk.org/centos but I'm not sure what the
expected time frame for that is.

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] DAHDI-linux-complete on CENTOS kernel 2.6.32

2011-08-14 Thread Patrick Lists 

On 08/14/2011 10:31 PM, Shaun Ruffell wrote:
[snip]

While I can't say I've run against that particular CENTOS kernel version, I
would be very surprised (and interested to know) if you had any problems.


Iirc that kernel (2.6.32-71.el6.i686) is a stock Red Hat Enterprise 
Linux 6 (or CentOS 6) kernel. Not sure if I understand you. Are you 
saying that Digium does not test their software on RHEL6/CentOS6 or that 
you tested it on RHEL6/CentOS6 but just not against the stock kernel?


Thanks,
Patrick

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[asterisk-users] Queue Breakout Input being Ignored

2011-08-14 Thread Anton Panetta 

Hello,

Raw stats:
Version:1.8.3.2
OS:Centos 5.6
Special setup: postgre database


I am having a few queue issues with Asterisk specifically relating to 
breaking out from queues while on hold.


The intent is that while someone is on hold they can press a key (lets 
say *) to break from the queue and go elsewhere (in this case to leave a 
message).


However In all of my testing I am unable to get this to work, or even to 
be able to leave the queue while on hold (to go to another queue for 
example, or force any action, eg hangup, play message, etc).


I have enabled DTMF debug and I can see in the console that asterisk is 
receiving the correct information, it just appears to be ignoring it.


Tested this on 1.6 as well and got the same results ( I had assumed it 
was a bug in 1.6) everything I look at suggests what I am doing -should- 
work.


I suspect I am missing some over arching setting or something incredibly 
simple but for the life of me I cannot get this function to work as 
described.


Here is some of the  config which I beleive is relevent.

Various variants have been tested with different actions to occur (sorry 
I dont have a simpler version of the below at hand)


[extensions.conf]

[app-helpdesk-bh]
exten => s,1(unanswered),Ringingsame  =>   n,Wait(2)
same  =>   n,Answer
same  =>   n,Wait(1)
same  =>   n(answered),NoOp(Helpdesk)
same  =>   n,Gosub(app-filteranon,s,1)
same  =>   n,Set(CALLERID(name)=${client} Helpdesk)
same  =>   n,NoOp(Caller ID set to: ${CALLERID(name)})
same  =>   n,NoOp(Callers waiting in queue: 
${QUEUE_WAITING_COUNT(helpdesk)})

; play the announcement for this helpdesk client (or the general intro)
same  =>   n(announceandwait),Gosub(app-helpdesk-${client}-intro,s,1)
same  =>   n,Gosub(app-helpdesk-${client}-special-rva,s,1)
; queue the call
;   t = allow the called party to transfer the call
;   k = allow the called party to park the call
;   c = continue on after call (to update devstate)
same  =>   n,Set(DEVICE_STATE(Custom:helpdesk)=RINGING)
same  =>   n,Queue(helpdesk,tk,,,600,,,app-update-helpdesk-queue-devstate)
same  =>   n,Gosub(app-update-helpdesk-queue-devstate,s,1)
; if the caller can't get into the queue or is kicked out of it, go to
; the after hours answering service
same  =>   n,Goto(app-helpdesk-hutchison,s,answered)
same  =>   n(done),NoOp

; update devstate after a hangup
exten => h,1,Gosub(app-update-helpdesk-queue-devstate,s,1)

include => app-helpdesk-breakout

[app-helpdesk-breakout]
exten => 1,1,Gosub(app-helpdesk-callback,s,1)
same  =>   n,Hangup

[app-helpdesk-callback]
; Callers brought to this context will be able to leave a message
; which will then be attached to an email and sent to the appropriate
; helpdesk RT queue
exten => s,1,NoOp(Helpdesk callback not implemented yet)
same  =>   n,Return
same  =>   n,GotoIf($["${CALLERID(num)}" = "anonymous" or 
${CALLERID(num)} = "0"]?collect)


same  =>   n(collect),Goto(app-helpdesk-callback-collect,s,1)

--
Anton Panetta
Application and Systems Administrator
NetSpot Pty. Ltd. ~ 183 Melbourne St. North Adelaide SA 5006
Ph: (+61 8) 7200 5333 | Fax: (+61 8) 8361 6811
Email: anton.pane...@netspot.com.au | Web: http://www.netspot.com.au
NetSpot - Enabling Technology for Education


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Re: [asterisk-users] Trouble with *8 Pickup

2011-08-14 Thread Alec Davis 
 

> > If you time the *8 just right so it is being handled during 
> the end of 
> > the Dial then I got:
> >
> > [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: 
> user_data 
> > is NULL [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: 
> > user_data is NULL
> 
> Does this happen when using the Pickup() application as well, 
> or is it specific to *8?
> 

Another way to get the same result, leaving orpaned channels, and the same
message as above.

Get the dialplan to execute
   Dial(SIP/phone1&SIP/phone2)

Then simultaneously dial *8 from 2 other phones in the same call/pickup
group as phone1 and phone2.

Alec Davis



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[asterisk-users] dynamically alter list of offered codecs (for faxing)

2011-08-14 Thread Norbert Zawodsky 

Hello everybody!

Lately I've had experiences that I'd like to share with you:

I did a some faxing over VOIP during the last two years. Not that much, 
lets say 1 fax per day on average. The setup is


Old analog fax machine <-> Linksys PAP2 ATA <-> Asterisk 1.2 <-> DSL <-> 
VoIP Provider 


I would estimate that 80% of the faxes went through on the first try. 
The rest aborted transmission with some communications error and needed 
a second (or rarely a 3rd) try.
Then suddenly, faxing didn't work that way any more. Every single fax 
needed many retries until it eventually went through.


Now, since I didn't change anything on my side I wondered what had 
happened. I enabled sip debug on the CLI an made a test fax.
I saw that my VoIP provider only offered codec alaw while the ATA was 
configured to only use ulaw.
So I assume that Asterisk had to perform some transcoding and maybe that 
broke the reliability ... ???
Since I didn't change anything on my side, the only reason I can think 
of ist that my provider changed some hardware or whatever and suddenly 
offers only alaw.
I reconfigured the ATA to only offer alaw and now every fax goes through 
on the first try without any problems.


Through this experience I had an idea:

The list of preferred codecs is "statically" set up in sip.conf.
Is it possible to modify that list dynamically in the dialplan for the 
outbound leg?
What I think of is to force the audio-stream to alaw für fax calls (= 
calls to/from a specific extension), but offer for example gsm for 
speech calls.


(For example: FAX machine is connected to etension 1234. If a call is 
made from extension 1234, Asterisk should offer only alaw to the 
_provider_ side)


Norbert



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Re: [asterisk-users] DID to display the calling number

2011-08-14 Thread Pezhman Lali 
some PRI providers block the callerid, and you can use only your head number
as cid.

best
Pezhman Lali
http://blog.lopl.net

On Sun, Aug 14, 2011 at 4:40 AM, Steve Edwards wrote:

> On Sat, 13 Aug 2011, bilal ghayyad wrote:
>
>  I need that if five IP Phones make outside calls, then destination should
>> see only 56725111...
>>
>
> You can set the caller ID before you dial.
>
> Whether it 'works' depends on your provider.
>
> Some providers will let you set the caller ID to anything, some will only
> allow you to set it to a number allocated to you, some will not let you set
> it at all.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
>
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>



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Re: [asterisk-users] DAHDI-linux-complete on CENTOS kernel 2.6.32

2011-08-14 Thread Shaun Ruffell 
On Mon, Aug 15, 2011 at 12:26:20AM +0800, Jesie Paluca wrote:
> 
> Does anyone had ever try to install DAHDI-tools and DAHI-linux on CENTOS
> kernel 2.6.32-71.el6.i686?  On the subversion source,  I can see the latest
> DAHDI-linux-complete version is 2.5.0+2.5.0.  Need any feedback if
> DAHDI-linux version 2.5.0 has any compatibility issue with linux kernel
> 2.6.  Apart from the DAHDI-linux-complete, understand that one of DAHDI
> dependency which is kernel source (kernel-debug-devel-2.6.32-71.el6.i686)
> needs to be installed as well.
> 
> Any inputs is highly appreciated.

While I can't say I've run against that particular CENTOS kernel version, I
would be very surprised (and interested to know) if you had any problems.

DAHDI-Linux runs fine under vanilla Linux 2.6.32.

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] callgroup more than 63

2011-08-14 Thread Thorolf Godawa 
Hi,

> is any problem if the max of callgroup >63, in source code ?
I set it to 1023 in Asterisk 1.4.20 and it seems to work without
problems on several servers.
-- 

Chau y hasta luego,

Thorolf

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[asterisk-users] Adhearsion 1.2.0 Released at Lone Star Ruby Conference V

2011-08-14 Thread Ben Klang 
Yesterday I announced the release of Adhearsion 1.2.0 during my presentation 
"State of the Art Telephony with Ruby" at Lone Star Ruby Conference V in 
Austin, Texas. LSRC has long been a supporter of the Adhearsion and Asterisk 
projects and I felt it fitting to use the presentation there as the opportunity 
to make this exciting new release.

In case you missed it, an overview of some of the new features was discussed in 
a previous blog post entitled "Upcoming Features in Adhearsion 1.2" on the Mojo 
Lingo blog. A quick recap here:

Native support for text-to-speech on both Asterisk and Tropo
Several useful new methods, including #play_or_speak, #record_to_file, #input!, 
#interruptible_play!, #speak and more.
Enhancements to existing methods such as #input and #play
Fixes to the way we load Bundler gems with the ahn executable
Improvments to the logging system to enable tracing the activities of a 
specific call through the logs
Also interesting to the Adhearsion community is the greatly improved support 
for Tropo via AGItate and the ability to easily create sophisticated 
text-to-speech renderings with the new RubySpeech library. These projects were
As always, the latest and greatest Adhearsion is available through RubyGems and 
sources are on Github.

Thanks once again to past core contributor Ben Langfeld and welcome to new 
contributor Lance Gleason. Thanks also to IfByPhone for sponsoring several of 
the improvements (especially Text-to-Speech).

/BAK/
-- 
Ben Klang
bkl...@mojolingo.com
404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @MojoLingo
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[asterisk-users] DAHDI-linux-complete on CENTOS kernel 2.6.32

2011-08-14 Thread Jesie Paluca 
Hi

Does anyone had ever try to install DAHDI-tools and DAHI-linux on CENTOS
kernel 2.6.32-71.el6.i686?  On the subversion source,  I can see the latest
DAHDI-linux-complete version is 2.5.0+2.5.0.  Need any feedback if
DAHDI-linux version 2.5.0 has any compatibility issue with linux kernel
2.6.  Apart from the DAHDI-linux-complete, understand that one of DAHDI
dependency which is kernel source (kernel-debug-devel-2.6.32-71.el6.i686)
needs to be installed as well.

Any inputs is highly appreciated.

Regards

Jesie
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing

2011-08-14 Thread Tzafrir Cohen 
On Fri, Aug 12, 2011 at 04:32:09PM +0100, Roger Burton West wrote:
> On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote:
> >Yes, same server, same filesystem...
> 
> I don't do Python, but a web search for shutil.move suggests that it
> doesn't reliably use the "rename" syscall. Might be worth shelling out
> to your system's mv command.

Or better: run it under strace and verify it does.

-- 
   Tzafrir Cohen
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http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] 1.6.2.20 ${DIALSTATUS} disagrees with CDR(answered)

2011-08-14 Thread Eric Wieling 

I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing.  
Below is a dialplan snippet and the resulting CLI output.  This is running in 
an 'h' extension.

Noop(DIALSTATUS=${DIALSTATUS})
Noop(CDR(disposition)=${CDR(disposition)})

-- Executing [h@pbxmax-dial-simple:1] NoOp("SIP/msx_01-005b", 
"DIALSTATUS=ANSWER") in new stack
-- Executing [h@pbxmax-dial-simple:2] NoOp("SIP/msx_01-005b", 
"CDR(disposition)=NO ANSWER") in new stack

Unless I seriously misunderstand the CDR(disposition) function, this looks like 
a bug to me.  Does any else have this issue?




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[asterisk-users] callgroup more than 63

2011-08-14 Thread Pezhman Lali 
Dear
is any problem if the max of callgroup >63, in source code ?
best

-- 
Pezhman Lali
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