Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Olivier]

2011/7/9 Doug Lytle 

> Can you say a Virtualized Asterisk with a PRI card!
>
> http://www.phoronix.com/scan.**php?page=news_item&px=OTY0OQ
>
> Doug
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
> --
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Great news !
Thanks for sharing !
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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Warren Selby]

On Fri, Jul 8, 2011 at 9:00 PM, Doug Lytle  wrote:
> Warren Selby wrote:
>>
>> Not trying to start a war here,
>
>
> That may be, but I have experience with VB.
>
> Doug


I use VB on my main desktop that runs windows in order to setup test
environments, so I understand.  :)

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Robert-iPhone]

+1 for Xen
-1 for VB


Sent from my iPhone

On Jul 8, 2011, at 10:00 PM, Doug Lytle  wrote:

> Warren Selby wrote:
>> Not trying to start a war here,
> 
> 
> That may be, but I have experience with VB.
> 
> Doug
> 
> 
> -- 
> Ben Franklin quote:
> 
> "Those who would give up Essential Liberty to purchase a little Temporary 
> Safety, deserve neither Liberty nor Safety."
> 
> 
> --
> _
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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Doug Lytle]

Warren Selby wrote:

Not trying to start a war here,



That may be, but I have experience with VB.

Doug


--
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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Warren Selby]

On Fri, Jul 8, 2011 at 6:28 PM, Doug Lytle  wrote:
> Can you say a Virtualized Asterisk with a PRI card!
>
> http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ
>
> Doug

Not trying to start a war here, but I thought Xen VM server has been
able to do this for a while now?

-- 
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--Warren Selby, dCAP
http://www.SelbyTech.com

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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Steve Totaro]

On Fri, Jul 8, 2011 at 8:12 PM, Doug Lytle  wrote:

> Patrick Lists wrote:
>
>> With virtualized environments prone to timing issues does this make sense
>> at all?
>>
>
> Wouldn't the timing be taken from the PCI device?  This is how it's working
> on my systems now.  Also, it's not something I would be put into production
> until I've hammered it.
>
>
> Doug
>
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
Hammer it.  Setup whatever requires the most timing, IAX, MOH, conference
bridge or whatever.

Setup SIPp (http://sipp.sourceforge.net/) to initiate x calls for the
duration and then you call in and see where it breaks.

Just to test the timing.

Piggy back a couple of quad pri systems and have them call each other and
SIPp into whatever app is most sensitive to timing.

what does cat /proc/interrupts show?

This is cool stuff, I will lab it up when I get some spare cycles.  I wonder
what happens with multiple VMs running Asterisk, I guess only one gets
access to the card, or it would be a mess.

Doesn't someone make a USB dongle for timing in VMs?  Anyone have input on
the best way get timing to multiple VMs.

I think it is time to start playing with Asterisk in various VM platforms
and configurations.  I want a stock underlying OS and stock guest OSes.

Xorcom has that USB FXX device.  Maybe a few of those, or multiple dongles.
Multiple FXX PCI cards with no modules assigned to different VMs?

Thanks,
Steve Totaro
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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Krishna Sumanth Chava]

Hi Doug,

May be you can try a PCI-E card that has a PRI port and asterisk on itself
and eliminating the need to install asterisk.

www.positrontelecom.com have these cards.

Thanks,
Krishna

On Fri, Jul 8, 2011 at 8:12 PM, Doug Lytle  wrote:

> Patrick Lists wrote:
>
>> With virtualized environments prone to timing issues does this make sense
>> at all?
>>
>
> Wouldn't the timing be taken from the PCI device?  This is how it's working
> on my systems now.  Also, it's not something I would be put into production
> until I've hammered it.
>
>
> Doug
>
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>  
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Doug Lytle]

Patrick Lists wrote:
With virtualized environments prone to timing issues does this make 
sense at all?


Wouldn't the timing be taken from the PCI device?  This is how it's 
working on my systems now.  Also, it's not something I would be put into 
production until I've hammered it.


Doug


--
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deserve neither Liberty nor Safety."


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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Doug Lytle]

Steve Totaro wrote:
Source code? 



http://download.virtualbox.org/virtualbox/4.1.0_BETA2/VirtualBox-4.1.0_BETA2.tar.bz2

Doug


--
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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Patrick Lists]

On 07/09/2011 01:28 AM, Doug Lytle wrote:

Can you say a Virtualized Asterisk with a PRI card!

http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ


With virtualized environments prone to timing issues does this make 
sense at all?


Regards,
Patrick

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Re: [asterisk-users] Issue 0019268 Patch Asterisk

[Patrick Lists]

On 07/08/2011 08:46 PM, Michael L. Young wrote:

Patrick,

The patch was merged in to the 1.8 branch on 5/13/2011 as revision 318783 
(http://svnview.digium.com/svn/asterisk?revision=318783&view=revision).

1.8.5-rc1 was tagged on 06/29/2011 
(http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1/?view=log).

So, it would be in there since the tag is copied from the 1.8 branch at 
revision 325707.

Looking at the code... it is definitely in there.  Line 5503 was moved to line 
5505.

Michael
(elguero)


Thanks for clearing that up Michael. I was looking at the wrong section.

Regards,
Patrick

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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Steve Totaro]

On Fri, Jul 8, 2011 at 7:28 PM, Doug Lytle  wrote:

> Can you say a Virtualized Asterisk with a PRI card!
>
> http://www.phoronix.com/scan.**php?page=news_item&px=OTY0OQ
>
> Doug
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
Source code?
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[asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Doug Lytle]

Can you say a Virtualized Asterisk with a PRI card!

http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ

Doug

--
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Re: [asterisk-users] FXO ports locking up

[Alec Davis]

> Is there a way to detect that there is no longer really an 
> active call happening and force a hangup or reset the 
> channel?  It'd be great if this could happen automatically.  
> Or as a temporary fix , is there a way to setup and extension 
> that the SIP phone could dial which would clear any active 
> calls associated with it?  Right now if this happens, I need 
> to login to the Asterisk CLI and issue a hangup command.  If 
> I don't, the channel appears to be in-use forever.

This may be the answer

sip.conf:

;--- RTP timers

; These timers are currently used for both audio and video streams. The RTP
timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
rtptimeout=60   ; Terminate call if 60 seconds of no RTP or
RTCP activity
; on the audio channel
; when we're not on hold. This is to be able
to hangup
; a call in the case of a phone disappearing
from the net,
; like a powerloss or grandma tripping over
a cable.

Alec Davis


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Re: [asterisk-users] DTMF issues still

[vmedina]

Latest firmware is  on the card

Sent from my android device.

-Original Message-
From: Jim Dickenson 
To: asterisk-users@lists.digium.com
Sent: Fri, 08 Jul 2011 5:59 PM
Subject: Re: [asterisk-users] DTMF issues still

I had a very strange problem with a Sangoma card that I had both Sangoma (about 
3 hours) and Digium (about 2 hours) look at. When I got a different Sangoma 
tech to look at the problem it went away. I told the tech he did something and 
he said I alway verify the firmware on the card is updated and as it was not I 
updated it. That fixed the problem.


This system had worked before a dahdi update was applied.


Bottom line make sure you have the most current firmware for your card.

-- 

Jim Dickenson

mailto:dicken...@cfmc.com


CfMC

http://www.cfmc.com/




On Jul 8, 2011, at 2:49 PM, vmed...@apcn.net wrote:


I am still having major issues with dtmf recognition. My setup is Polycom end 
points. Tried this with different models, firmware and cfgs. Outbound calls are 
not going out reliably. Phones are set to rfc2833. I have had sangoma and 
elastix support look at it.. No better. Running asterisk 1.8.4. What am I 
missing? Any help appreciated.. Btw.. Used a basic test set and dialed out on 
all lines no problem. Sangoma card is a a400 with echo cancel.


Sent from my android device.

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Re: [asterisk-users] DTMF issues still

[Jim Dickenson]

I had a very strange problem with a Sangoma card that I had both Sangoma (about 
3 hours) and Digium (about 2 hours) look at. When I got a different Sangoma 
tech to look at the problem it went away. I told the tech he did something and 
he said I alway verify the firmware on the card is updated and as it was not I 
updated it. That fixed the problem.

This system had worked before a dahdi update was applied.

Bottom line make sure you have the most current firmware for your card.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jul 8, 2011, at 2:49 PM, vmed...@apcn.net wrote:

> I am still having major issues with dtmf recognition. My setup is Polycom end 
> points. Tried this with different models, firmware and cfgs. Outbound calls 
> are not going out reliably. Phones are set to rfc2833. I have had sangoma and 
> elastix support look at it.. No better. Running asterisk 1.8.4. What am I 
> missing? Any help appreciated.. Btw.. Used a basic test set and dialed out on 
> all lines no problem. Sangoma card is a a400 with echo cancel.
> 
> 
> Sent from my android device.
> 
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[asterisk-users] DTMF issues still

[vmedina]

I am still having major issues with dtmf recognition. My setup is Polycom end 
points. Tried this with different models, firmware and cfgs. Outbound calls are 
not going out reliably. Phones are set to rfc2833. I have had sangoma and 
elastix support look at it.. No better. Running asterisk 1.8.4. What am I 
missing? Any help appreciated.. Btw.. Used a basic test set and dialed out on 
all lines no problem. Sangoma card is a a400 with echo cancel.


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[asterisk-users] Cisco ATA 187 configuration file

[William Muriithi]

Hi all,

I have sincerely spent the last 2 hours googling for ATA.cnf.xml
file for a Cisco ATA 187 and I am not going anywhere.  Either no one
is using this thing out there or it has worked very well for them.

Would there be someone of this list using the TFTP managed  Cisco ATA
187 with Asterisk?  I would be very grateful if you can share a copy
of the file with all the confidential information stripped out and
replaced with a place holder

Do have a great weekend and thanks in advance

Regards,

Wiulliam

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Re: [asterisk-users] How to create a module

[Miguel Molina]

El 08/07/11 12:50, Steve Edwards escribió:
*) You can execute hundreds of AGIs written in C in the time it takes 
to load the Perl interpreter and parse your script. 

Just curious... have you timed this to demonstrate?
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Re: [asterisk-users] check_auth: username mismatch

[Mike Diehl]

"Dan Journo"  wrote:
>> I've got a Polycom 501 that I just can't seem to get lines 2 and 3 to
work on.
>>  Line 1 works fine.
> 
> Last time I had that issue, it resolved itself when i restarted Asterisk.

Any ideas as to what causes it, though?

> Are you able to do that?

I'm scheduling a SIP reload tonight.  Might as well do an Asterisk restart
instead. 

--

Take care and have fun,
Mike Diehl.



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[asterisk-users] Problems with DTMF Caller ID

[Antonio Modesto]

Good afternoon,

I'm trying to configure my asterisk to work with DTMF signaling (I
live in Brazil) , i've put these lines in my chan_dahdi.conf

usecallerid=yes
callerid=asreceived
cidsignaling=dtmf
cidstart=polarity

my dahdi system.conf

loadzone=br
defaultzone=br
fxsks=1
fxsks=3
fxsks=4

When i try to dial to my asterisk box via analog line, it returns
these errors in the console:

  == Starting post polarity CID detection on channel 2
-- Starting simple switch on 'DAHDI/2-1'
[Jul  8 14:33:02] WARNING[2889]: sig_analog.c:2482 __analog_ss_thread:
Channel DAHDI/2-1 in prering state, but I have nothing to do.
Terminating simple switch, should be restarted by the actual ring.
-- Hanging up on 'DAHDI/2-1'
-- Hungup 'DAHDI/2-1'

Information about my system:
-> FreeBSD 8.2-STABLE
-> asterisk-1.8.4.2
-> dahdi-2.4.0rc5_5
-> Digium TDM400P Board with 3 FXO Modules
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Re: [asterisk-users] Issue 0019268 Patch Asterisk

[Michael L. Young]

- Original Message -
> From: "Patrick Lists" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, July 8, 2011 1:58:36 PM
> Subject: Re: [asterisk-users] Issue 0019268 Patch Asterisk
> 
> On 07/08/2011 07:32 PM, Mark Rosedale wrote:
> 
> > * channels/sig_pri.c: PRI early media won't ring. And another way
> >   to pass early media. Don't indicate that there is inband
> >   information present, just assume that the B channel is
> >   connected.
> >   * Restore clearing the dialing flag Rx squelch unconditionally
> >   when a PROCEEDING message comes in. (closes issue #19268)
> >   Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by
> >   rmudgett (license 664) Tested by: tbsky
> 
> I looked at the 1.8.5-rc1 code. I don't see that patch in the
> 1.8.5-rc1
> branch. Did I miss something?
> 
> >> http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1/channels/sig_pri.c?view=markup
> 
> Regards,
> Patrick
> 

Patrick,

The patch was merged in to the 1.8 branch on 5/13/2011 as revision 318783 
(http://svnview.digium.com/svn/asterisk?revision=318783&view=revision).

1.8.5-rc1 was tagged on 06/29/2011 
(http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1/?view=log).

So, it would be in there since the tag is copied from the 1.8 branch at 
revision 325707.

Looking at the code... it is definitely in there.  Line 5503 was moved to line 
5505.

Michael
(elguero)

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Re: [asterisk-users] How to create a module

[Adrian Abramovich]

Steve,

Yes, I'm executing the Perl script as an AGI.
Very useful points! I really appreciate them!.

Thanks,


Adrian Abramovich

On Fri, Jul 8, 2011 at 12:50 PM, Steve Edwards wrote:

> On Fri, 8 Jul 2011, Adrian Abramovich wrote:
>
>  We are using asterisk 1.4 and we use a Perl script to record some specific
>> calls. As far, everything is working well. I was thinking about create a
>> module in order to improve script's performance.
>>
>
> (I'm assuming you're executing your Perl script as an AGI.)
>
> I'd vote for re-writing your Perl script in C.
>
> Once you get past the small* hit for process creation and AGI environment
> setup, the execution time should be very similar between C code executing as
> a module and C code executing as an AGI.
>
> The advantages of an AGI over a module are:
>
> ) You already know the AGI environment.
>
> ) Writing an AGI is probably an order of magnitude easier than writing a
> module. The 'environment' is much simpler, stable between Asterisk versions
> and better defined.
>
> ) Debugging an AGI is probably a couple of orders of magnitude easier than
> debugging a module because it is usually a single threaded process and you
> can (within obvious limitations) 'desk test' your code completely external
> to Asterisk.
>
> ) You'll get more support on the mailing lists. A lot more people know how
> to write and debug an AGI than know how to write and debug a module.
>
> ) When (not if) you hit a bug, it only takes out your AGI, not the entire
> Asterisk process. The damage is limited to a single call, not all calls in
> flight.
>
> *) You can execute hundreds of AGIs written in C in the time it takes to
> load the Perl interpreter and parse your script.
>
> --
> 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] Issue 0019268 Patch Asterisk

[Patrick Lists]

On 07/08/2011 07:32 PM, Mark Rosedale wrote:


* channels/sig_pri.c: PRI early media won't ring. And another way
  to pass early media. Don't indicate that there is inband
  information present, just assume that the B channel is connected.
  * Restore clearing the dialing flag Rx squelch unconditionally
  when a PROCEEDING message comes in. (closes issue #19268)
  Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by
  rmudgett (license 664) Tested by: tbsky


I looked at the 1.8.5-rc1 code. I don't see that patch in the 1.8.5-rc1 
branch. Did I miss something?



http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1/channels/sig_pri.c?view=markup


Regards,
Patrick

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Re: [asterisk-users] How to create a module

[Adrian Abramovich]

Thanks Steve!... I will try the tip and I will let you know the result.

Adrian Abramovich

On Fri, Jul 8, 2011 at 12:26 PM, Steve Murphy  wrote:

>
>
> On Fri, Jul 8, 2011 at 10:39 AM, Adrian Abramovich <
> adrianabramov...@gmail.com> wrote:
>
>> Hi,
>>
>> We are using asterisk 1.4 and we use a Perl script to record some specific
>> calls. As far, everything is working well.
>> I was thinking about create a module in order to improve script's
>> performance.
>> I checked the Russell's blog:
>>
>> http://www.russellbryant.net/blog/2008/06/19/how-to-write-an-asterisk-module-part-1/
>> This is a old post and I would like to know if there are something new.
>>
>
> What I do, is look at the other apps/funcs for guidance. Pick the smallest
> first,
> and you can copy their style and layout. The module spec has evolved from
> 1.4
> to 1.6 to 1.8, but it's the same basics (to a degree).
>
>
>> Is it a good idea to move to module?
>>
>
> If it increases performance, and you need that, then heck yes! The only
> drawback is that
> it *is* in the source; you'll have to tweak it as you move up the versions.
> You have to compile
> and install it. Perl would remain static (I would imagine) even when you
> update asterisk.
>
>
>> Thanks in advance,
>>
>> Adrian Abramovich
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Steve Murphy
>
> ParseTree Corporation
>
> 57 Lane 17
>
> Cody, WY 82414
>
> ✉  m...@parsetree.com
>
> ☎ 307-899-5535
>
>
>
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Re: [asterisk-users] How to create a module

[Steve Edwards]

On Fri, 8 Jul 2011, Adrian Abramovich wrote:

We are using asterisk 1.4 and we use a Perl script to record some 
specific calls. As far, everything is working well. I was thinking about 
create a module in order to improve script's performance.


(I'm assuming you're executing your Perl script as an AGI.)

I'd vote for re-writing your Perl script in C.

Once you get past the small* hit for process creation and AGI environment 
setup, the execution time should be very similar between C code executing 
as a module and C code executing as an AGI.


The advantages of an AGI over a module are:

) You already know the AGI environment.

) Writing an AGI is probably an order of magnitude easier than writing a 
module. The 'environment' is much simpler, stable between Asterisk 
versions and better defined.


) Debugging an AGI is probably a couple of orders of magnitude easier than 
debugging a module because it is usually a single threaded process and you 
can (within obvious limitations) 'desk test' your code completely external 
to Asterisk.


) You'll get more support on the mailing lists. A lot more people know how 
to write and debug an AGI than know how to write and debug a module.


) When (not if) you hit a bug, it only takes out your AGI, not the entire 
Asterisk process. The damage is limited to a single call, not all calls in 
flight.


*) You can execute hundreds of AGIs written in C in the time it takes to 
load the Perl interpreter and parse your script.


--
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] Issue 0019268 Patch Asterisk

[Mark Rosedale]

From the change log
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1

2011-05-13 16:28 + [r318783-318868]  Richard Mudgett 

* main/features.c: CDR's are being written immediately on caller
  hangup. CDR's are being written immediately on caller hangup. The
  dialplan is not able to modify it in the h exten. The h exten in
  the initial context is not run before closing CDR's when the
  bridge is unlinked if a macro is active and does not have an h
  exten. * Make ast_bridge_call() check for an h exten in the
  current context and if a macro is active then the initial
  context. The first h exten found is then run before closing the
  CDR. (closes issue #18212) Reported by: leearcher Patches:
  issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested
  by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/

* channels/sig_pri.c: PRI early media won't ring. And another way
  to pass early media. Don't indicate that there is inband
  information present, just assume that the B channel is connected.
  * Restore clearing the dialing flag Rx squelch unconditionally
  when a PROCEEDING message comes in. (closes issue #19268)
  Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by
  rmudgett (license 664) Tested by: tbsky

On Jul 8, 2011, at 12:42 PM, Patrick Lists wrote:

> On 07/08/2011 06:07 PM, Mark Rosedale wrote:
>> Looks like the patch is in 1.8.5-rc1...I may just roll with that version. If 
>> that doesn't work then I may patch it manually like you suggest.
> 
> Not sure where you looked but afaict that patch has not been applied to 
> 1.8.5-rc1:
> 
> http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1/channels/sig_pri.c?view=markup
> 
> See line 5399 and 5416.
> 
> Regards,
> Patrick
> 
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Re: [asterisk-users] How to create a module

[Steve Murphy]

On Fri, Jul 8, 2011 at 10:39 AM, Adrian Abramovich <
adrianabramov...@gmail.com> wrote:

> Hi,
>
> We are using asterisk 1.4 and we use a Perl script to record some specific
> calls. As far, everything is working well.
> I was thinking about create a module in order to improve script's
> performance.
> I checked the Russell's blog:
>
> http://www.russellbryant.net/blog/2008/06/19/how-to-write-an-asterisk-module-part-1/
> This is a old post and I would like to know if there are something new.
>

What I do, is look at the other apps/funcs for guidance. Pick the smallest
first,
and you can copy their style and layout. The module spec has evolved from
1.4
to 1.6 to 1.8, but it's the same basics (to a degree).


> Is it a good idea to move to module?
>

If it increases performance, and you need that, then heck yes! The only
drawback is that
it *is* in the source; you'll have to tweak it as you move up the versions.
You have to compile
and install it. Perl would remain static (I would imagine) even when you
update asterisk.


> Thanks in advance,
>
> Adrian Abramovich
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

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✉  m...@parsetree.com

☎ 307-899-5535
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Re: [asterisk-users] Issue 0019268 Patch Asterisk

[Patrick Lists]

On 07/08/2011 06:07 PM, Mark Rosedale wrote:

Looks like the patch is in 1.8.5-rc1...I may just roll with that version. If 
that doesn't work then I may patch it manually like you suggest.


Not sure where you looked but afaict that patch has not been applied to 
1.8.5-rc1:


http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1/channels/sig_pri.c?view=markup

See line 5399 and 5416.

Regards,
Patrick

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[asterisk-users] How to create a module

[Adrian Abramovich]

Hi,

We are using asterisk 1.4 and we use a Perl script to record some specific
calls. As far, everything is working well.
I was thinking about create a module in order to improve script's
performance.
I checked the Russell's blog:
http://www.russellbryant.net/blog/2008/06/19/how-to-write-an-asterisk-module-part-1/
This is a old post and I would like to know if there are something new.
Is it a good idea to move to module?
Thanks in advance,

Adrian Abramovich
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[asterisk-users] Asterisk meetme and Timer ?

[Thomas Elsgaard]

Hi

I have just downloaded AsteriskNow, and i would like to use the MeetMe
conference, but i can in varoius postings read that i need some
timing? Is this really needed? And how do i setup this on the
AsteriskNow 1.7 installation ??

I am using the AsteriskNow for pure VoIP, no PSTN interworking.

Thomas

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Re: [asterisk-users] Issue 0019268 Patch Asterisk

[Mark Rosedale]

Looks like the patch is in 1.8.5-rc1...I may just roll with that version. If 
that doesn't work then I may patch it manually like you suggest.

Thanks,
mjr
On Jul 8, 2011, at 12:00 PM, Patrick Lists wrote:

> On 07/08/2011 05:01 PM, Mark Rosedale wrote:
> 
>> This is not working the source code of 1.8.4.4. I assume that the patch
>> is for a different version. Any ideas about how to apply this patch to
>> 1.8.4.4 so that I can avoid using the svn branch?
> 
> It's a 2 line patch. If you look at the source it's easy to spot why the 
> patch failed (some code was added). Line 5399 is removed and the addition 
> comes before line 5417:
> 
> http://svnview.digium.com/svn/asterisk/branches/1.8/channels/sig_pri.c?view=markup
> 
> Why don't you try to apply it manually?
> 
> Regards,
> Patrick
> 
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Re: [asterisk-users] Issue 0019268 Patch Asterisk

[Patrick Lists]

On 07/08/2011 05:01 PM, Mark Rosedale wrote:


This is not working the source code of 1.8.4.4. I assume that the patch
is for a different version. Any ideas about how to apply this patch to
1.8.4.4 so that I can avoid using the svn branch?


It's a 2 line patch. If you look at the source it's easy to spot why the 
patch failed (some code was added). Line 5399 is removed and the 
addition comes before line 5417:


http://svnview.digium.com/svn/asterisk/branches/1.8/channels/sig_pri.c?view=markup

Why don't you try to apply it manually?

Regards,
Patrick

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Re: [asterisk-users] Problem in Detecting Dtmf on FXO line.

[Ruben Rögels]

Am 08.07.2011 08:58, schrieb DHAVAL INDRODIYA:
> Hi All,
> 
> I am having Problem in detecting DTMF on analog lines. basically are
> system is in india and telco provider is BSNL [Bharat sanchar Nigam
> LImited].
> 
> We have Purchased Analog card From chinaroby.com 
> which is X1600P 16 port FXO  card. they also provide us wctdm.c file.
> 
> card is detected successfully, incoming and outgoing calls scenario is
> also fine.
> 
> we are unable to receive dtmf properly it means there is some digit are
> missing when we receive dtmf the ratio of sucess is about to 70% and 30%
> of calls are getting wrong dtmf .
> 
> Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24
> 
> I load module using
> modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1
> fixedtimepolarity=16
> 
> here id  chan_dahdi.conf.

Hello,

did you try plaing with rxgain and txgain?
When I set up a TDM400, I had some issues with DTMF because the signals
where overmodulated.

Regards,
Ruben

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[asterisk-users] Issue 0019268 Patch Asterisk

[Mark Rosedale]

So in troubleshooting a different issue I'm having I decided to upgrade one of 
my backup servers to 1.8.4.4 instead of running on the svn branch. Ultimately 
this is where I'd like to be in the end. 

However, I'm having one issue that seems to be related to this ticket 
(https://issues.asterisk.org/view.php?id=19268). 

Any outbound long distance call has no ring. Basically you hear silence until 
the phone is connected. This was one of the things that pushed us to the svn 
branch in the first place. However, I noticed that they have a patch. 

wget 'https://issues.asterisk.org/file_download.php?file_id=29388&type=bug' -O 
- | patch -p0

This is not working the source code of 1.8.4.4. I assume that the patch is for 
a different version. Any ideas about how to apply this patch to 1.8.4.4 so that 
I can avoid using the svn branch?

This is the error I'm getting 
patching file channels/sig_pri.c
Hunk #1 FAILED at 5500.
1 out of 1 hunk FAILED -- saving rejects to file channels/sig_pri.c.rej
 

Thanks,
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[asterisk-users] FXO ports locking up

[Shawn L]

I have a situation where I have an Asterisk box which receives 8
analog lines from a
Mitel PBX and then drives 8 cordless SIP phones in a 1-to-1 mapping (a
call coming in
on port 1 of the digium FXO board is delivered to SIP phone 1, an
outgoing call on SIP
phone 2 goes out FXO line 2, etc.

This works fine normally, but every once in a while (no set time, or
pattern that I can
see -- It may be caused by the wifi sip phone going out of range of an
access point and
not coming back into range fast enough) the FXO port does not hangup
after the call is
terminated and just sits in an in-use state.  Since it's a 1-to-1
mapping, the SIP phone
associated with the in-use line now produces a fast busy when you
attempt to make a
call because it cannot get an outbound line.

Is there a way to detect that there is no longer really an active call
happening and force a
hangup or reset the channel?  It'd be great if this could happen
automatically.  Or as a
temporary fix , is there a way to setup and extension that the SIP
phone could dial which
would clear any active calls associated with it?  Right now if this
happens, I need to login
to the Asterisk CLI and issue a hangup command.  If I don't, the
channel appears to be
in-use forever.

Thanks

Shawn


The setup is fairly straight-forward

Extensions
[in-phone2]
exten => s,1,Answer()
exten => s,n,Noop(CALLERID(name))
exten => s,n,Noop(CALLERID(num))
exten => s,n,Dial(SIP/cordless2,25,tTo)
exten => s,n,Hangup

[out-phone2]
exten => _[*#0-9]!,1,Dial(${LINE2}/${EXTEN})
exten => _[*#0-9]!,2,Congestion()
exten => _[*#0-9]!,102,Congestion()

[cordless2]
type=friend
qualify=yes
rtptimeout=1
secret=
call-limit=1
nat=no
host=dynamic
canreinvite=no
context=out-phone2
callerid="cordless2" <102>

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Re: [asterisk-users] Anybody doing PRI over IP?

[Shawn L]

>
> Right, this is how I expected it to operate. My prior question though was 
> regarding the 'T1 over Ethernet' scheme someone mentioned which ran full 
> throughput all the time.
>

That is true.  If you're doing a clear-channel or pseudo-wire T1 over
ethernet you will always be using 1.54 Mbps weather the T1 has any
data on it or not.

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Re: [asterisk-users] timeout with outbound calls

[Eric Wieling]

Show us the CLI output of the failed call.

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> salaheddine elharit
> Sent: Friday, July 08, 2011 10:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] timeout with outbound calls
>
> i have tested this solution and i have the same issue
>
> in my case want to call a phone number 06 from my
> snom phone (sip223)
>
> the issue still the same
>
> any help please
>
>
> 2011/7/8 Eric Wieling 
>
>
>
>
>   > -Original Message-
>   > From: asterisk-users-boun...@lists.digium.com
>   > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>   > salaheddine elharit
>   > Sent: Friday, July 08, 2011 6:43 AM
>   > To: Asterisk Users Mailing List - Non-Commercial Discussion
>   > Subject: [asterisk-users] timeout with outbound calls
>
>   >
>   > Hi
>   >
>   > i want to use timeout  with asterisk 1.4 in order to hangup
>   > the outbound calls after 25 sec
>   >
>   > i call my mobile number 067xxx from my sip acount 223
>   > and i want to hangu up the call automatic after 25 sec  but
>   > there is no hangup after 25
>   >
>   > could you please help me
>   >
>   > exten => 223,1,Set(TIMEOUT(absolute)=25) exten =>
>   > 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
>   > exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
>   > exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
>   > exten => 223,n,Hangup();
>   >
>   > Best Regards.
>   >
>
>
>   pbx*CLI> core show application dial
>
>-= Info about application 'Dial' =-
>
>   [Synopsis]
>   Attempt to connect to another device or endpoint and
> bridge the call.
>   [snip]
>  L(x[:y[:z]]):
>  x - Maximum call time, in milliseconds
>  y - Warning time, in milliseconds
>  z - Repeat time, in milliseconds
>  Limit the call to  milliseconds. Play a warning
> when  mill
>  iseconds are left. Repeat the warning every 
> milliseconds until time
>  expires.
>  This option is affected by the following variables:
>  ${LIMIT_PLAYAUDIO_CALLER}:
>  yes
>  no
>  If set, this variable causes Asterisk to play the
>  prompts to the caller.
>  ${LIMIT_PLAYAUDIO_CALLEE}:
>  yes
>  no
>  If set, this variable causes Asterisk to play the
>  prompts to the callee.
>  ${LIMIT_TIMEOUT_FILE}:
>  filename
>  If specified,  specifies the sound prompt
>  to play when the timeout is reached. If not
> set, the time remaining
>  will be announced.
>  ${LIMIT_CONNECT_FILE}:
>  filename
>  If specified,  specifies the sound prompt
>  to play when the call begins. If not set,
> the time remaining will
>  be announced.
>  ${LIMIT_WARNING_FILE}:
>  filename
>  If specified,  specifies the sound prompt
>  to play as a warning when time  is
> reached. If not set, the
>  time remaining will be announced.
>   [snip]
>
>
>   --
>
> _
>   -- Bandwidth and Colocation Provided by
> http://www.api-digital.com   --
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> webinar every Thurs:
> http://www.asterisk.org/hello
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>

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Re: [asterisk-users] timeout with outbound calls

[salaheddine elharit]

i have tested this solution and i have the same issue

in my case want to call a phone number 06 from my snom phone
(sip223)

the issue still the same

any help please

2011/7/8 Eric Wieling 

>
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > salaheddine elharit
> > Sent: Friday, July 08, 2011 6:43 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] timeout with outbound calls
>  >
> > Hi
> >
> > i want to use timeout  with asterisk 1.4 in order to hangup
> > the outbound calls after 25 sec
> >
> > i call my mobile number 067xxx from my sip acount 223
> > and i want to hangu up the call automatic after 25 sec  but
> > there is no hangup after 25
> >
> > could you please help me
> >
> > exten => 223,1,Set(TIMEOUT(absolute)=25) exten =>
> > 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> > exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> > exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
> > exten => 223,n,Hangup();
> >
> > Best Regards.
> >
>
> pbx*CLI> core show application dial
>
>  -= Info about application 'Dial' =-
>
> [Synopsis]
> Attempt to connect to another device or endpoint and bridge the call.
> [snip]
>L(x[:y[:z]]):
>x - Maximum call time, in milliseconds
>y - Warning time, in milliseconds
>z - Repeat time, in milliseconds
>Limit the call to  milliseconds. Play a warning when  mill
>iseconds are left. Repeat the warning every  milliseconds until time
>expires.
>This option is affected by the following variables:
>${LIMIT_PLAYAUDIO_CALLER}:
>yes
>no
>If set, this variable causes Asterisk to play the
>prompts to the caller.
>${LIMIT_PLAYAUDIO_CALLEE}:
>yes
>no
>If set, this variable causes Asterisk to play the
>prompts to the callee.
>${LIMIT_TIMEOUT_FILE}:
>filename
>If specified,  specifies the sound prompt
>to play when the timeout is reached. If not set, the time
> remaining
>will be announced.
>${LIMIT_CONNECT_FILE}:
>filename
>If specified,  specifies the sound prompt
>to play when the call begins. If not set, the time remaining
> will
>be announced.
>${LIMIT_WARNING_FILE}:
>filename
>If specified,  specifies the sound prompt
>to play as a warning when time  is reached. If not set, the
>time remaining will be announced.
> [snip]
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] timeout with outbound calls

[Umair Bari]

You may also use

exten => 223,n,Dial(SIP/${EXTEN},,KktL(25000))


On Fri, Jul 8, 2011 at 5:33 PM, Eric Wieling  wrote:

>
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > salaheddine elharit
> > Sent: Friday, July 08, 2011 6:43 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] timeout with outbound calls
> >
> > Hi
> >
> > i want to use timeout  with asterisk 1.4 in order to hangup
> > the outbound calls after 25 sec
> >
> > i call my mobile number 067xxx from my sip acount 223
> > and i want to hangu up the call automatic after 25 sec  but
> > there is no hangup after 25
> >
> > could you please help me
> >
> > exten => 223,1,Set(TIMEOUT(absolute)=25) exten =>
> > 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> > exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> > exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
> > exten => 223,n,Hangup();
> >
> > Best Regards.
> >
>
> pbx*CLI> core show application dial
>
>  -= Info about application 'Dial' =-
>
> [Synopsis]
> Attempt to connect to another device or endpoint and bridge the call.
> [snip]
>L(x[:y[:z]]):
>x - Maximum call time, in milliseconds
>y - Warning time, in milliseconds
>z - Repeat time, in milliseconds
>Limit the call to  milliseconds. Play a warning when  mill
>iseconds are left. Repeat the warning every  milliseconds until time
>expires.
>This option is affected by the following variables:
>${LIMIT_PLAYAUDIO_CALLER}:
>yes
>no
>If set, this variable causes Asterisk to play the
>prompts to the caller.
>${LIMIT_PLAYAUDIO_CALLEE}:
>yes
>no
>If set, this variable causes Asterisk to play the
>prompts to the callee.
>${LIMIT_TIMEOUT_FILE}:
>filename
>If specified,  specifies the sound prompt
>to play when the timeout is reached. If not set, the time
> remaining
>will be announced.
>${LIMIT_CONNECT_FILE}:
>filename
>If specified,  specifies the sound prompt
>to play when the call begins. If not set, the time remaining
> will
>be announced.
>${LIMIT_WARNING_FILE}:
>filename
>If specified,  specifies the sound prompt
>to play as a warning when time  is reached. If not set, the
>time remaining will be announced.
> [snip]
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

Thanks & Regards,

Umair Bari
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Re: [asterisk-users] Master.csv file limit

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup
Sent: Friday, July 08, 2011 8:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Master.csv file limit

 

What is the maximum size limit of Master.csv file and what happens when it
reaches limit?

 

Thanks

I don't believe there is a specified size limit on Master.csv.  your local
ulimit settings might cut it off at a certain size, otherwise the only
constrain should be available disk space.  What would happen when the limit
is reached is that calls would not be written to this file and possibly the
running instance of Asterisk would shut down.   The best "prevention" for
this would be to "roll" the file periodically.  There are several wikis that
explain this.  

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Re: [asterisk-users] Master.csv file limit

[Gergo Csibra]

Friday, July 8, 2011, 3:13:01 PM, deeps wrote:

> What is the maximum size limit of Master.csv file and what happens when it
> reaches limit?

That is a text file. Only limited by filesystem.

-- 
Best regards,
 Gergomailto:csi...@gmail.com


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[asterisk-users] Master.csv file limit

[deeps backup]

What is the maximum size limit of Master.csv file and what happens when it
reaches limit?


Thanks
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Re: [asterisk-users] dialout time configuration

[Satish Barot]

What do you mean by ring time out?
See the 'timeout' in Queue application. keep it blank if you just want to
keep your callers in queue for infinite time.
Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule[,position])

[SATISH]
Mumbai, India


On Fri, Jul 8, 2011 at 3:53 PM, Deka, Rajib IN MAA SL <
rajib.d...@siemens.com> wrote:

>  Hi List,
>
>
>
> Is it possible to configure an infinite ring timeout for queue in asterisk?
>
> I mean, the caller should be able to be in queue until and unless he
> disconnects the call.
>
>
>
> Thanks,
>
> Rajib
>
> --
> Important notice: This e-mail and any attachment there to contains
> corporate proprietary information. If you have received it by mistake,
> please notify us immediately by reply e-mail and delete this e-mail and its
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> Thank You.
>
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Re: [asterisk-users] timeout with outbound calls

[Eric Wieling]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> salaheddine elharit
> Sent: Friday, July 08, 2011 6:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] timeout with outbound calls
>
> Hi
>
> i want to use timeout  with asterisk 1.4 in order to hangup
> the outbound calls after 25 sec
>
> i call my mobile number 067xxx from my sip acount 223
> and i want to hangu up the call automatic after 25 sec  but
> there is no hangup after 25
>
> could you please help me
>
> exten => 223,1,Set(TIMEOUT(absolute)=25) exten =>
> 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
> exten => 223,n,Hangup();
>
> Best Regards.
>

pbx*CLI> core show application dial

  -= Info about application 'Dial' =-

[Synopsis]
Attempt to connect to another device or endpoint and bridge the call.
[snip]
L(x[:y[:z]]):
x - Maximum call time, in milliseconds
y - Warning time, in milliseconds
z - Repeat time, in milliseconds
Limit the call to  milliseconds. Play a warning when  mill
iseconds are left. Repeat the warning every  milliseconds until time
expires.
This option is affected by the following variables:
${LIMIT_PLAYAUDIO_CALLER}:
yes
no
If set, this variable causes Asterisk to play the
prompts to the caller.
${LIMIT_PLAYAUDIO_CALLEE}:
yes
no
If set, this variable causes Asterisk to play the
prompts to the callee.
${LIMIT_TIMEOUT_FILE}:
filename
If specified,  specifies the sound prompt
to play when the timeout is reached. If not set, the time remaining
will be announced.
${LIMIT_CONNECT_FILE}:
filename
If specified,  specifies the sound prompt
to play when the call begins. If not set, the time remaining will
be announced.
${LIMIT_WARNING_FILE}:
filename
If specified,  specifies the sound prompt
to play as a warning when time  is reached. If not set, the
time remaining will be announced.
[snip]

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Re: [asterisk-users] timeout with outbound calls

[A J Stiles]

On Friday 08 Jul 2011, salaheddine elharit wrote:
> what can i do in order to fix this issue

If and when an absolute timeout occurs, Asterisk jumps to the "T" extension.  
So, in the same context as your 223 extension, you need something like

exten => T,1,NoOp(Absolute timeout triggered)
exten => T,n,Hangup()

This will write "Absolute timeout triggered" to the console and hang up.  Put 
whatever commands in there you like, obviously .

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] timeout with outbound calls

[salaheddine elharit]

what can i do in order to fix this issue

regards

2011/7/8 A J Stiles 

>  On Friday 08 Jul 2011, salaheddine elharit wrote:
> > i want to use timeout  with asterisk 1.4 in order to hangup the outbound
> > calls after 25 sec
> >
> > i call my mobile number 067xxx from my sip acount 223  and i want to
> > hangu up the call automatic after 25 sec  but there is no hangup after 25
> >
> > could you please help me
> >
> > exten => 223,1,Set(TIMEOUT(absolute)=25)
> > exten => 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> > exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> > exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
> > exten => 223,n,Hangup();
>
> What have you got in your "T" extension?  When the absolute timeout
> expires,
> it will jump here.
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] dialout time configuration

[Faisal Hanif]

I think yes. Check queuetimout variable.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deka, Rajib IN
MAA SL
Sent: Friday, July 08, 2011 3:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dialout time configuration

 

Hi List,

 

Is it possible to configure an infinite ring timeout for queue in asterisk?

I mean, the caller should be able to be in queue until and unless he
disconnects the call.

 

Thanks,

Rajib

 

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Re: [asterisk-users] timeout with outbound calls

[A J Stiles]

On Friday 08 Jul 2011, salaheddine elharit wrote:
> i want to use timeout  with asterisk 1.4 in order to hangup the outbound
> calls after 25 sec
>
> i call my mobile number 067xxx from my sip acount 223  and i want to
> hangu up the call automatic after 25 sec  but there is no hangup after 25
>
> could you please help me
>
> exten => 223,1,Set(TIMEOUT(absolute)=25)
> exten => 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
> exten => 223,n,Hangup();

What have you got in your "T" extension?  When the absolute timeout expires, 
it will jump here.

-- 
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Answers come *after* questions.

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[asterisk-users] timeout with outbound calls

[salaheddine elharit]

Hi

i want to use timeout  with asterisk 1.4 in order to hangup the outbound
calls after 25 sec

i call my mobile number 067xxx from my sip acount 223  and i want to
hangu up the call automatic after 25 sec  but there is no hangup after 25

could you please help me

exten => 223,1,Set(TIMEOUT(absolute)=25)
exten => 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
exten => 223,n,Hangup();

Best Regards.
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[asterisk-users] dialout time configuration

[Deka, Rajib IN MAA SL]

Hi List,

Is it possible to configure an infinite ring timeout for queue in asterisk?
I mean, the caller should be able to be in queue until and unless he 
disconnects the call.

Thanks,
Rajib


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Re: [asterisk-users] avoid indicate condition 9 and starting music on hold

[Alexandre Rodrigues]

Hello Giedrius,

Same problem with me in asterisk 1.4 using a dahdi module. I am trying
to transfer a call using
flash hook but I get the same message.

Did you resolved this issue?

Thanks in advance,

Alex

2009/8/18 Giedrius Augys :
> Hello,
>
>   I've a problem. I've asterisk 1.6.0.5 version. And I've created
> callcenter, but agents registers to another SIP server. When agent tries
> transfer a client to another operator , pressing flash, I get this:
> [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know
> how to indicate condition 9
> [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data: Unable to
> handle indication 9 for 'SIP/xxx.xxx.xx-082b9c80'
>     -- Started music on hold, class 'default', on SIP/sip.call.lt-082b9c80
> Is it possible to avoid this? I don't want that in this situation (after
> pressing Flash), asterisk starts Musing On Hold.
>
> Thanks for help
> --
> Pagarbiai  / Best Regards,
> Giedrius
>
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Re: [asterisk-users] Help for Alcatel asterisk

[Alexandre Rodrigues]

Hello Carlos,

I have the same problem when I try to do a flash hook with dahdi module.
Did you resolve your problem?


Thanks in advance,

Alex

2009/8/13 Carlos Rojas :
> Hello everybody
>
> I have an asterisk with an integration of alcatel pbx, by sip trunk, all
> calls are fine, but tha calls calls that originate from a analog line,
> the recipient is not listening, and that if they hear the call originates,
> the lines are E1 in alcatel pbx.
>
> When a asteris user call to analog line the call is ok.
>
>
> Everyone, has been that problem?
>
> I change asterisk version 1.4.21 to 1.4.18 but the same problem.
>
> I saw  the cli
>
> [Aug 12 16:15:40] WARNING[2997]: chan_sip.c:3927 sip_indicate: Don't know
> how to indicate condition 9
> [Aug 12 16:15:40] WARNING[2997]: channel.c:2369 ast_indicate_data: Unable to
> handle indication 9 for 'SIP/4001-0a16f5c0'
>
> Anyone can help me..
>
>
> Regards
>
> ___
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[asterisk-users] DB Driven IVR

[G M]

I am using Vicidial and I am looking for someone who can help with DB Driven
IVR.
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[asterisk-users] asterisk and eyebeam

[salaheddine elharit]

Hello

With x-lite version 3.0 I can limited the time of call in



Option===>advanced===>network RTCP has been inactive for 30sec (by default)



with Eyebeam 1.5.19 i do the same but the call has not been hang-up



If there is any way to do that with Eyebeam or in extensions.conf



Thanks and regards
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[asterisk-users] HI

[David @ULC]

hI
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Re: [asterisk-users] Problem in Detecting Dtmf on FXO line.

[DHAVAL INDRODIYA]

Yes dear i have tried diable also with yes and no. but no successful result
found/

On Fri, Jul 8, 2011 at 12:41 PM, Faisal Hanif  wrote:

> Did u tried by disabling relaxdtmf?
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
> *Sent:* Friday, July 08, 2011 11:58 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Problem in Detecting Dtmf on FXO line.
>
> ** **
>
> Hi All,
>
> I am having Problem in detecting DTMF on analog lines. basically are system
> is in india and telco provider is BSNL [Bharat sanchar Nigam LImited].
>
> We have Purchased Analog card From chinaroby.com which is X1600P 16 port
> FXO  card. they also provide us wctdm.c file.
>
> card is detected successfully, incoming and outgoing calls scenario is also
> fine.
>
> we are unable to receive dtmf properly it means there is some digit are
> missing when we receive dtmf the ratio of sucess is about to 70% and 30% of
> calls are getting wrong dtmf .
>
> Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24
>
> I load module using
> modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1
> fixedtimepolarity=16
>
> here id  chan_dahdi.conf.
>
> [trunkgroups]
>
> [channels]
> context=from-zaptel
> signalling=fxs_ks
> busydetect=yes
> busycount=4
> ;rxwink=300  ; Atlas seems to use long (250ms) winks
> usecallerid=yes
> callerid=asreceived
> cidstart=polarity_in
> cidsignalling=dtmf
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> callprogess=yes
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800
> rxgain=0.0
> txgain=0.0
> ;cid_rxgain=5.0
> relaxdtmf=yes
> callgroup=1
> pickupgroup=1
> toneduration=500
> ;answeronpolarityswitch=yes
> hanguponpolarityswitch=yes
> ;polarityonanswerdelay=1000
>
> group=0
> channel => 1
> ;channel => 2
> ;channel => 3
> ;channel => 4
> ;channel => 5
> ;channel => 6
> ;channel => 7
> ;channel => 8
> ;channel => 9
> ;channel => 10
> ;channel => 11
> ;channel => 12
> ;channel => 13
> ;channel => 14
> ;channel => 15
> ;channel => 16
>
>
> Also set tonezone = in in system.conf, tried many solutions and changed so
> many parameters of chan_dahdi.cong but still i am not getting successful
> result.
>
>
> Please share your comments if anyone have idea for india specific region .
>
> Regards
> Dhaval
>
> 
>
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Re: [asterisk-users] Problem in Detecting Dtmf on FXO line.

[Faisal Hanif]

Did u tried by disabling relaxdtmf?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Friday, July 08, 2011 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem in Detecting Dtmf on FXO line.

 

Hi All,

I am having Problem in detecting DTMF on analog lines. basically are system
is in india and telco provider is BSNL [Bharat sanchar Nigam LImited].

We have Purchased Analog card From chinaroby.com which is X1600P 16 port FXO
card. they also provide us wctdm.c file.

card is detected successfully, incoming and outgoing calls scenario is also
fine.

we are unable to receive dtmf properly it means there is some digit are
missing when we receive dtmf the ratio of sucess is about to 70% and 30% of
calls are getting wrong dtmf .

Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24

I load module using 
modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1
fixedtimepolarity=16

here id  chan_dahdi.conf.

[trunkgroups] 
 
[channels] 
context=from-zaptel 
signalling=fxs_ks 
busydetect=yes 
busycount=4 
;rxwink=300  ; Atlas seems to use long (250ms) winks 
usecallerid=yes 
callerid=asreceived 
cidstart=polarity_in 
cidsignalling=dtmf 
hidecallerid=no 
callwaiting=yes 
usecallingpres=yes 
callwaitingcallerid=yes 
threewaycalling=yes 
transfer=yes 
canpark=yes 
cancallforward=yes 
callreturn=yes 
callprogess=yes 
echocancel=yes 
echocancelwhenbridged=no 
echotraining=800 
rxgain=0.0 
txgain=0.0 
;cid_rxgain=5.0 
relaxdtmf=yes 
callgroup=1 
pickupgroup=1 
toneduration=500 
;answeronpolarityswitch=yes 
hanguponpolarityswitch=yes 
;polarityonanswerdelay=1000 
 
group=0 
channel => 1 
;channel => 2 
;channel => 3 
;channel => 4 
;channel => 5 
;channel => 6 
;channel => 7 
;channel => 8 
;channel => 9 
;channel => 10 
;channel => 11 
;channel => 12 
;channel => 13 
;channel => 14 
;channel => 15 
;channel => 16


Also set tonezone = in in system.conf, tried many solutions and changed so
many parameters of chan_dahdi.cong but still i am not getting successful
result.


Please share your comments if anyone have idea for india specific region .

Regards
Dhaval



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