[asterisk-users] Asterisk 8 and mixmonitor

2012-05-02 Thread ik 
Hello,

I have weird issue with Asterisk 8 lately.

When I call MixMonitor without mixing the channels, it changes the
sides of in and out.
Sometimes the first leg of the call is in and sometimes it's out.

I can't figure out if it's a known issue, or a new bug.

I'm using Asterisk 8.11.1

Any ideas how can I figure out what is leg is what file ?

Thanks,
Ido

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Re: [asterisk-users] Asterisk 8 and mixmonitor

2012-05-02 Thread ik 
On Wed, May 2, 2012 at 3:44 PM, SamyGo govoi...@gmail.com wrote:
 I can't figure out if it's a known issue, or a new bug.


 Or a new feature !!

 Can you share the dialplan code where you are executing the mixmon
 application !

I use it using the manager:

DEBUG ami 2012-05-02 15:43:53  Sending AMI action:
 Action: monitor
 ActionID: pZWLT4gi-3OFk-emUX-6xiu-4GptK0BXJXuR
 Channel: Local/leg_a@some-context-9710;2
 File: /var/spool/asterisk/recordings/wav/121e0009a9327900c0b9b3d5d5db7426
 Mix: 0


(I edited the number and context name)



 Regards,
 Sammy.


 On Wed, May 2, 2012 at 5:09 PM, ik ido...@gmail.com wrote:

 Hello,

 I have weird issue with Asterisk 8 lately.

 When I call MixMonitor without mixing the channels, it changes the
 sides of in and out.
 Sometimes the first leg of the call is in and sometimes it's out.

 I can't figure out if it's a known issue, or a new bug.

 I'm using Asterisk 8.11.1

 Any ideas how can I figure out what is leg is what file ?

 Thanks,
 Ido

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[asterisk-users] Dahdi, PRI and all circuits are busy now

2012-02-06 Thread ik 
Hello,

I have a weird case, when some numbers dialed using a PRI, have an early
media sounds instead of normal ringing.
Few of the numbers are making Asterisk 1.6 (using Elastix 2) to report all
circuits are busy now. All of this numbers are cellular phones, but they
constantly reporting the same thing.
But most numbers even with such music, works well.

What do you think I should look for, to figure out if that's a bug, or
something else in the Asterisk side ?

Thanks,
Ido
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Re: [asterisk-users] Fax not detected by Asterisk

2011-11-18 Thread ik 
On Fri, Nov 18, 2011 at 04:01, Edwin Lam edwin@officegeneral.comwrote:

 On 11/17/11 3:30 AM, ik wrote:


 I have an Elastix 2 machine with digium fax modules (with license).
 When I try to create an extension that also works with FAX, Asterisk does
 not
 detect any incoming fax. Even when I use 'fax set debug', it does not
 display
 anything.
 It's Asterisk 6.2.x . Any ideas what can I do to figure out why it does
 not detect
 the arriving faxes ?


 in your dialplan, did you Answer the call and Wait
 a few seconds for Asterisk to detect the fax tone before
 you do some other things?


Yes, I placed a wait up to 10 seconds just to make sure




 --
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 Systems Engineer, OfficeWyze, Inc.
 Ph: +1 415 439 4988 Fax: +1 415 283 3370
 http://pgpkeys.mit.edu:11371/**pks/lookup?op=getsearch=**0xD6506D20http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20



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[asterisk-users] Fax not detected by Asterisk

2011-11-17 Thread ik 
Hello List,

I have an Elastix 2 machine with digium fax modules (with license).
When I try to create an extension that also works with FAX, Asterisk does
not detect any incoming fax. Even when I use 'fax set debug', it does not
display anything.
It's Asterisk 6.2.x . Any ideas what can I do to figure out why it does not
detect the arriving faxes ?

Thanks,

Ido
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Re: [asterisk-users] Fax not detected by Asterisk

2011-11-17 Thread ik 
On Thu, Nov 17, 2011 at 16:31, Danny Nicholas da...@debsinc.com wrote:

 Fax is detected at ring/start level.  Check chan_dahdi.conf.


The chan_dahdi.conf contain faxdetect=both


 

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *ik
 *Sent:* Thursday, November 17, 2011 5:30 AM
 *To:* Asterisk Users Mailing List
 *Subject:* [asterisk-users] Fax not detected by Asterisk

 ** **

 Hello List,

 I have an Elastix 2 machine with digium fax modules (with license).
 When I try to create an extension that also works with FAX, Asterisk does
 not detect any incoming fax. Even when I use 'fax set debug', it does not
 display anything.
 It's Asterisk 6.2.x . Any ideas what can I do to figure out why it does
 not detect the arriving faxes ?

 Thanks,

 Ido

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Re: [asterisk-users] [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression

2011-08-18 Thread ik 
I'm using it.

Can you please provide more information on the issue with this feature ?
Is there another way to know the response code of SIP ?

Thanks,

Ido

On Thu, Aug 18, 2011 at 15:42, Matthew Nicholson mnichol...@digium.comwrote:

 Greetings,

 Recently a performance regression in chan_sip was discovered in Asterisk
 1.8. The regression is caused by chan_sip setting
 MASTER_CHANNEL(HASH(SIP_CAUSE,chan name)) after each response received
 on a channel. That feature has been made optional in the latest 1.8 SVN
 code, but is currently still enabled by default. After some internal
 discussion, we decided to consider disabling this feature by default in
 future 1.8 versions. This would be an unexpected behavior change for
 anyone depending on that SIP_CAUSE update in their dialplan.
 Alternatively, with this feature enabled, anyone upgrading from Asterisk
 1.4 will see a 60% decrease in the amount of SIP traffic they can handle
 before encountering problems.

 Before disabling this feature, we wanted to get a feel for how many
 people are using it. If you use this feature, please respond to this
 email and let us know.
 --
 Matthew Nicholson
 Digium, Inc. | Software Developer




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Re: [asterisk-users] ${HASH(SIP_CAUSE, ...)} and peer name

2011-07-12 Thread ik 
On Tue, Jul 12, 2011 at 00:22, Philippe Sultan philippe.sul...@gmail.comwrote:

 The destination channel dies right after your Dial statement exits,
 but you can retrieve the info in the channel that's still alive :
 exten = _XXX,n,Dial(SIP/${EXTEN})
 exten = _XXX,n,NoOp(SIP return code :
 ${HASH(SIP_CAUSE,${CDR(dstchannel)})})

 Works fine on the Asterisk server I'm running (1.8.3.3).


Thanks, that works for me as well.



 Philippe


Ido



 On Mon, Jul 11, 2011 at 11:01 PM, ik ido...@gmail.com wrote:
  Hello,
 
  I'm trying to figure out what was the return code of SIP for a call.
  The problem is that HASH(SIP_CAUSE) require a peer name, but when I try
 to
  retrieve the peer name using ${CHANNEL(peername)}, I have an error
 message
  that CHANNEL does not have peername or it is not available to be used.
  I tried to print it with NOOP on a live channel, and also after hangup,
 both
  with the same error message.
 
  So how can I get SIP_CAUSE, or how can I get the peer name ?
 
  Thanks,
 
  Ido
 
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[asterisk-users] ${HASH(SIP_CAUSE, ...)} and peer name

2011-07-11 Thread ik 
Hello,

I'm trying to figure out what was the return code of SIP for a call.
The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to
retrieve the peer name using ${CHANNEL(peername)}, I have an error message
that CHANNEL does not have peername or it is not available to be used.
I tried to print it with NOOP on a live channel, and also after hangup, both
with the same error message.

So how can I get SIP_CAUSE, or how can I get the peer name ?

Thanks,

Ido
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Re: [asterisk-users] Asterisk and Archlinux

2010-04-24 Thread ik 
On Sat, Apr 24, 2010 at 21:19, Christian christia...@runbox.com wrote:

 Hi all,
 Is anyone here using Asterisk on Archlinux?


Yes and no, I do use it on Archlinux for testing purpose but not as a
server.
Arch linux is not built to be a server distro, unlike Debian that have extra
steps for process handling, like restarting a service that was just updated
and more.


 If so, was it much to do in order for it to work?


You need to either use the AUR builds or download the ABS information and
build it for yourself. Personally I use the yaourt tool as a pacman front
end.


 Do you also use Dahdi?


Like any other Asterisk it must have a dahdi module, at least dahdi_dummy.


 many thanks,
 Christian


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Ido
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[asterisk-users] MeetMe and setting conference timeout

2009-06-01 Thread ik 
Hello,

I have MeetMe rooms generated dynamically and it always have two people
inside that are entered by dialplan.

I wish to make in some way a timeout mechanism that after X amount of time,
it will disconnect the users and kick them out of the conference.
How can I do such thing ?

Thanks,

Ido
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Re: [asterisk-users] MeetMe and setting conference timeout

2009-06-01 Thread ik 
Thank you all for the comments.

Jared, I've implemented your idea, and it worked very well. Thank you very
much for it :)

Ido


On Mon, Jun 1, 2009 at 10:36 PM, Jared Smith jsm...@digium.com wrote:

 On Mon, 2009-06-01 at 10:22 -0500, Danny Nicholas wrote:
  Write an AGI to hangup the users using Asterisk Manager.  If you’re
  the ambitious type, you could do it with grep and awk from the
  dialplan; just hangup the appropriate channels.

 Wow... that sure sounds like complicated overkill to me.  Why not just
 set an absolute timeout on the channels?  Something like:

 exten = 123,1,Set(TIMEOUT(absolute)=3600)
 exten = 123,n,MeetMe(blah,d)

 --
 Jared Smith
 Training Manager
 Digium, Inc.


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[asterisk-users] manager ignore my settings

2008-02-20 Thread ik 
Hello,

I have the following settings for manager on two Asterisk 1.2.24 (that
have installed over a year ago):

[user]
secret = password
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
write = call,command

On one server, Asterisk only react as you would expect - sending a
command without having any verbose on anything. On the other machine,
I have verbose like I enabled everything, including the read option.

Another issue I have on the weird machine is that I have unexplained
crash (where safe asterisk return asterisk to life), the core dump
each time is different, but one thing is in common: it all fails on a
free command.

Any ideas what might cause this issues, and what should I be looking for ?

Thanks,
Ido Kanner
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