Re: [asterisk-users] CallerID shows wrong values in manager interface

[Devraj Mukherjee]

Thanks all :)

Appreciate it.

On Feb 1, 2008 12:04 PM, Ex Vito <[EMAIL PROTECTED]> wrote:
>   I've struggled with this recently. In short:
>
>
>   - Observed behaviour is expected as of asterisk 1.2 and later,
> as previously described by Mojo
>
>   - If you want to get the caller id for the channel calling (dialling)
> into that channel for that specific Newstate: Ringing event, you
> can use the 'o' flag to the Dial command; in this case you'll get
> old asterisk 1.0 behaviour -- do you really want to depend on
> such an old behaviour ? well I decided I didn't...
>
>   - Otherwise, you'll need to track other events (IIRC, at least, Dial,
> AgentCalled, Newstate, etc) in the AMI so as to know who is calling
> who at a given instant
>
>   - BEWARE: if memory serves me right (search the list archives in the Nov/Dec
> timeframe), the behaviour is not 100% homogeneous for different channel
> types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from
> one channel to the other is that a) at times you get the Dial
> event first then the
> Newstate: Ringing event; and that b) with other/different
> orig/dest channel types
> you'll get the events in the reverse order... Nothing much but: i)
> you'll have to
> track them either way and ii) it reveals that the AMI events
> aren't 100% clean!!!
>
>   :/
> --
>   exvito
>
>
> On Feb 1, 2008 12:08 AM, Mojo with Horan & Company, LLC
> <[EMAIL PROTECTED]> wrote:
> > The snippet is asterisk telling you "I'm just letting you know that the
> > correct caller id for Channel: SIP/103-098500d8 is CallerID: 103"
> >
> > This is absolutely correct, it's just not a piece of information you
> > expected to be receiving at that point.
> >
> > You probably also received a packet like that with the following:
> >  Channel: SIP/101-
> >  CallerID: 101
> > telling you, again, the caller id for only that channel.
> >
> > Moj
> >
>
>
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-- 
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Incredibles)

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[asterisk-users] call log notice messages

[Vieri]

I am getting quite a lot of these notices:

Feb  1 10:22:01 NOTICE[2487] cdr.c: CDR on channel
'mISDN/3-2' not posted
Feb  1 10:22:01 NOTICE[2487] cdr.c: CDR on channel
'mISDN/3-2' lacks end
Feb  1 10:22:06 NOTICE[2471] cdr.c: CDR on channel
'mISDN/3-1' not posted
Feb  1 10:22:06 NOTICE[2471] cdr.c: CDR on channel
'mISDN/3-1' lacks end
Feb  1 10:22:17 NOTICE[1591] cdr.c: CDR on channel
'mISDN/4-1' not posted
Feb  1 10:22:17 NOTICE[1591] cdr.c: CDR on channel
'mISDN/4-1' lacks end
Feb  1 10:23:40 NOTICE[2744] cdr.c: CDR on channel
'mISDN/3-1' not posted
Feb  1 10:23:40 NOTICE[2744] cdr.c: CDR on channel
'mISDN/3-1' lacks end
Feb  1 10:23:41 NOTICE[2157] cdr.c: CDR on channel
'mISDN/2-1' not posted
Feb  1 10:23:41 NOTICE[2157] cdr.c: CDR on channel
'mISDN/2-1' lacks end
Feb  1 10:23:51 NOTICE[2386] cdr.c: CDR on channel
'Zap/15-1' not posted
Feb  1 10:23:51 NOTICE[2386] cdr.c: CDR on channel
'Zap/15-1' lacks end
Feb  1 10:25:44 NOTICE[2516] cdr.c: CDR on channel
'mISDN/2-2' not posted
Feb  1 10:25:44 NOTICE[2516] cdr.c: CDR on channel
'mISDN/2-2' lacks end
Feb  1 10:26:23 NOTICE[3064] cdr.c: CDR on channel
'mISDN/4-1' not posted
Feb  1 10:26:23 NOTICE[3064] cdr.c: CDR on channel
'mISDN/4-1' lacks end
Feb  1 10:26:38 NOTICE[3086] cdr.c: CDR on channel
'SIP/4052-b519c460' not posted
Feb  1 10:26:38 NOTICE[3086] cdr.c: CDR on channel
'SIP/4052-b519c460' lacks end
Feb  1 10:26:50 NOTICE[3187] cdr.c: CDR on channel
'mISDN/4-2' not posted
Feb  1 10:26:50 NOTICE[3187] cdr.c: CDR on channel
'mISDN/4-2' lacks end
Feb  1 10:27:04 NOTICE[2857] cdr.c: CDR on channel
'mISDN/3-2' not posted
Feb  1 10:27:04 NOTICE[2857] cdr.c: CDR on channel
'mISDN/3-2' lacks end

What do they mean?
Should I worry about call details not being correctly
recorded?

Vieri



  

Never miss a thing.  Make Yahoo your home page. 
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[asterisk-users] meetme music on hold - when only conference member problem

[Tomasz Zieleniewski]

Hi,

I have the following problem that when someone connects to my conference and
is the only member
music on hold is played just for one second or less and then stops:

[Feb  1 10:38:46] -- Started music on hold, class 'default', on channel
'SIP/sip.touk.pl-0083dad0'
[Feb  1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0

I use the following command to invoke
exten => s,n,MeetMe(|cdIMps)

Kind regards
tomasz
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Re: [asterisk-users] h priority problem

[Rizwan Hisham]

You can use account code and userfield. You can set userfield to anything
you want in the dialplan.

On Feb 1, 2008 3:31 PM, Doug Lytle <[EMAIL PROTECTED]> wrote:

> Paul Hales wrote:
> >
> >
> > Anyone have any ideas? What can I use to carry a variable over into
> > 'h'??
> >
>
> Lets see what you have so far.
>
> 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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-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Re: [asterisk-users] h priority problem

[Doug Lytle]

Paul Hales wrote:
>   
>
> Anyone have any ideas? What can I use to carry a variable over into
> 'h'??
>   

Lets see what you have so far.

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] meetme music on hold - when only conference member problem

[Gordon Henderson]

On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote:

> Hi,
>
> I have the following problem that when someone connects to my conference and
> is the only member
> music on hold is played just for one second or less and then stops:
>
> [Feb  1 10:38:46] -- Started music on hold, class 'default', on channel
> 'SIP/sip.touk.pl-0083dad0'
> [Feb  1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0
>
> I use the following command to invoke
> exten => s,n,MeetMe(|cdIMps)

You probably don't have a timing source.

If you don't have any telephony hardware installed you'll need the ztdummy 
module...

Gordon


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Re: [asterisk-users] h priority problem

[Atis Lezdins]

On 2/1/08, Paul Hales <[EMAIL PROTECTED]> wrote:
>
> I need to carry a variable over into the 'h' priority - so I can go back
> and clean up DB entries in a mysql database (time of call and so on)
>
> I tried using UNIQUEID but it seems that 'h' generates a new one.
>
> Anyone have any ideas? What can I use to carry a variable over into
> 'h'??

You have to understand that there's several channels for one call.
Variables are only inherited from parent channel to child, if you make
them inheritable by prepending underscore or two underscores. For
example - if you want to have one common unique identifier for call -
do this at beginning of each channel:

if ("${call_id}"="") {
  Set(__call_id=${UNIQUEID});
}

Then you can use this to store something shared in asterisk internal DB()

Regards,
Atis



-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

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[asterisk-users] Unicall

[Mark Welch]

Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel
1.4.5.1.  If we were to update or recompile Asterisk, would we need to
do anything with Unicall or Zaptel?

 

Thanks in advance 

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[asterisk-users] play promt at the same time to calling and callee

[Giedrius Augys]

Hello,

   I want  that,  when call is answered , callee and calling would hear
different prompts and after promts the calls would be bridged. I've tried
this situation:
exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world)
exten =>
s,2,Dial(SIP/trunk-out/37052390920|60|rL(10)A(conf-enteringno))

But these prompts play not in the same time: just after conf-enteringno
prompt asterisk plays hello world promt.
--  Playing 'conf-enteringno' (language 'en')
 --  Playing 'hello-world' (language 'en')

So my question is , how to do this in the same time. Maybe somebody is using
Dial G(context^exten^pri) for this purpose?

Thanks
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Re: [asterisk-users] Asterisk-Addons install success-Could not find ooh323.conf

[Russell Bryant]

[EMAIL PROTECTED] wrote:
> Am I doing something wrong? What I should do to get ooh323.conf

cp asterisk-ooh323c/h323.conf.sample /etc/asterisk/ooh323.conf

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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Re: [asterisk-users] play promt at the same time to calling and callee

[Giedrius Augys]

2008/2/1, Giedrius Augys <[EMAIL PROTECTED]>:
>
> Hello,
>
>I want  that,  when call is answered , callee and calling would hear
> different prompts and after promts the calls would be bridged. I've tried
> this situation:
> exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world)
> exten =>
> s,2,Dial(SIP/trunk-out/37052390920|60|rL(10)A(conf-enteringno))
>
> But these prompts play not in the same time: just after conf-enteringno
> prompt asterisk plays hello world promt.
> --  Playing 'conf-enteringno' (language 'en')
>  --  Playing 'hello-world' (language 'en')
>
> So my question is , how to do this in the same time. Maybe somebody is
> using Dial G(context^exten^pri) for this purpose?
>
> Thanks
>


I have tried this :
exten => s,1,Dial(SIP/trunk-out/37052390920|60|rG(music-testinis^s^1))

[music-testinis]
exten => s,1,goto(1,1)
exten => s,2,goto(2,1)


exten => 1,1,Playback(lt/conf-enteringno)
exten => 2,1,Playback(lt/conf-enteringno)

but I get this:
god*CLI>
-- Executing [EMAIL PROTECTED]:1] Goto("SIP/sip3.call.lt-08141e00",
"testuojame|s|1") in new stack
-- Goto (testuojame,s,1)
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/sip3.call.lt-08141e00",
"SIP/trunk-out/37052390920|60|rG(music-testinis^s^1)") in new stack
-- Called trunk-out/37052390920
-- SIP/trunk-out-0818fb40 is ringing
-- SIP/trunk-out-0818fb40 is making progress passing it to
SIP/sip3.call.lt-08141e00
-- SIP/trunk-out-0818fb40 is making progress passing it to
SIP/sip3.call.lt-08141e00
-- SIP/trunk-out-0818fb40 answered SIP/sip3.call.lt-08141e00
-- Executing [EMAIL PROTECTED]:1] Goto("SIP/sip3.call.lt-08141e00",
"1|1") in new stack
-- Goto (music-testinis,1,1)
-- Executing [EMAIL PROTECTED]:1] Playback("SIP/sip3.call.lt-08141e00",
"lt/conf-enteringno") in new stack
--  Playing 'lt/conf-enteringno' (language
'en')
-- Executing [EMAIL PROTECTED]:2] Goto("SIP/trunk-out-0818fb40", "2|1")
in new stack
-- Goto (music-testinis,2,1)
-- Executing [EMAIL PROTECTED]:1] Playback("SIP/trunk-out-0818fb40",
"lt/conf-enteringno") in new stack
--  Playing 'lt/conf-enteringno' (language 'en')
  == Auto fallthrough, channel 'SIP/sip3.call.lt-08141e00' status is
'UNKNOWN'
  == Auto fallthrough, channel 'SIP/trunk-out-0818fb40' status is 'UNKNOWN'


My question is , how to bridge these two calls. I'm using Asterisk 1.4.11,
Thanks
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Re: [asterisk-users] BRI card with PCI-E interface

[Bob Pierce]

On Fri, 2008-02-01 at 15:01 +0100, Alberto Pastore wrote:
> Olivier ha scritto:
> > Hi,
> > 
> > Does such card exist ?
> > It seems all existing models are designed for PCI buses.
> > 
> > Regards
> > 
> > 
> > 
> > 
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> 
> The Sangoma A500BRX is a 2 to 6 bri pci-x interface,
> although I've never tested it
> (A500BRECX comes with hw echo cancellation).
> 

We've been using the Sangoma A104DE in production for almost a year now
and it works great.

Bob

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Re: [asterisk-users] realtime warning

[Rilawich Ango]

yes.

On Feb 1, 2008 12:07 PM, Russell Bryant <[EMAIL PROTECTED]> wrote:
>
> Rilawich Ango wrote:
> > Hi,
> > The server log shows the following message.
> >
> > [Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for
> > 'sippeers' found to engine 'mysql', but the engine is not available
> >
> > Does it mean the server failed to file the mysql server?  I have
> > installed mysql and both asterisk and mysql are located in the same
> > server.  What do the message mean?  It seems the message will cause
> > the user failed to login.  How can it be solved?
>
> Did you install res_config_mysql from asterisk-addons?
>
> --
> Russell Bryant
> Senior Software Engineer
> Open Source Team Lead
> Digium, Inc.
>
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[asterisk-users] Enterprise or Fedora?

[love U . all]

i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be 
more stable than Fedora core Linux  or it makes no significant difference
_
Express yourself instantly with MSN Messenger! Download today it's FREE!
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Re: [asterisk-users] BRI card with PCI-E interface

[Alberto Pastore]

Olivier ha scritto:
> Hi,
> 
> Does such card exist ?
> It seems all existing models are designed for PCI buses.
> 
> Regards
> 
> 
> 
> 
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The Sangoma A500BRX is a 2 to 6 bri pci-x interface,
although I've never tested it
(A500BRECX comes with hw echo cancellation).

Regards,
Alberto.

-- 
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it

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[asterisk-users] BRI card with PCI-E interface

[Olivier]

Hi,

Does such card exist ?
It seems all existing models are designed for PCI buses.

Regards
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Re: [asterisk-users] Problem with DTMF dialing

[Anthony Messina]

On Thursday 31 January 2008 11:52:09 pm Ian wrote:
> Sorry for taking so long to reply,
>
> This email got lost in translation, again.
>
> Ian
>
> Ian said the following on 30-Jan-08 03:57 PM
>
> > Thaks for the speedy reply
> >
> > Tzafrir Cohen said the following on 30-Jan-08 12:37 PM:
> >> On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote:
> >>> Hi all
> >>>
> >>> I have a small problem here. I asked this question on another asterisk
> >>> mailing list, but nobody seemed to be able to help me there.
> >>>
> >>> We are running
> >>>
> >>>* Asterisk 1.4.17
> >>>* Libpri 1.4.3
> >>>* Zaptel 1.4.8

did you use zaptel-1.4.7 prior to this?  did it work then?  if so, it may be 
related to http://bugs.digium.com/view.php?id=11855

-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] Enterprise or Fedora?

[Matthew J. Roth]

love U.all wrote:
> i wanna build a production Asterisk box ,will RedHat Linux Enterprise 
> Server be more stable than Fedora core Linux  or it makes no 
> significant difference
I started out running Fedora, but I have migrated away from it for a few 
reasons.  Fedora has a very short life cycle and since the demise of the 
legacy project there is no support for releases that are over a year 
old.  It is also considered a bleeding-edge distribution and is a 
testing ground for new applications and features.  As such, it has 
frequent updates (including the kernel) that aren't guaranteed not to 
break your system.

If stability and long-term support are your goals, I recommend taking a 
look at CentOS .  It is a binary-compatible 
clone of Red Hat Enterprise Linux that's free and has a very long 
support period.  It's basically RHEL without the paid support contract.  
My migration to CentOS was painless, because the file system and 
configuration are practically identical to those of Fedora.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer



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Re: [asterisk-users] Meetme voice quality problems

[Administrator TOOTAI]

Matthew J. Roth a écrit :
> [...]
>
> I settled on using an empty TDM400P as a timing source, because it is a 
> simple solution that "just works."  This may still be your best bet, but 
> I'll defer judgment on that to the list because Asterisk has evolved 
> quite a bit since I made that decision.
This is not true if you're using B410P cards. We always face timing 
problem as we can't -Asterisk stability issues- add X100P or TDM400P 
with those cards.

-- 
Daniel

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Re: [asterisk-users] meetme music on hold - when only conference member problem

[Tomasz Zieleniewski]

I have,
I have ztdummy module loaded in the kernel

On Feb 1, 2008 11:59 AM, Gordon Henderson <[EMAIL PROTECTED]>
wrote:

> On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote:
>
> > Hi,
> >
> > I have the following problem that when someone connects to my conference
> and
> > is the only member
> > music on hold is played just for one second or less and then stops:
> >
> > [Feb  1 10:38:46] -- Started music on hold, class 'default', on
> channel
> > 'SIP/sip.touk.pl-0083dad0'
> > [Feb  1 10:38:46] -- Stopped music on hold on
> SIP/sip.touk.pl-0083dad0
> >
> > I use the following command to invoke
> > exten => s,n,MeetMe(|cdIMps)
>
> You probably don't have a timing source.
>
> If you don't have any telephony hardware installed you'll need the ztdummy
> module...
>
> Gordon
>
>
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Re: [asterisk-users] AgentLogin by console

[equis software]

Thanks Tzafir, but this functionality needs sombody answer the call.
I need to do this automatically.



On Jan 22, 2008 4:10 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> On Tue, Jan 22, 2008 at 04:04:29PM -0200, equis software wrote:
> > Hi!
> > Is there any way to login an agent by console command?
> >
> > I want to login an agent doing this system call.
> >
> > asterisk -rx 'AgentCallbackLogin 304 [EMAIL PROTECTED]'
> >
> > Any ideas, thanks.
>
> Something of the sort of:
>
>  originate SIP/peer-to-login application AgentCallbackLogin
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Enterprise or Fedora?

[LWATCDR]

We are using Fedora because that is what the company we got our system
from recommended.
If I was doing the system myself I would throw in my vote for CentOS.
I am using it for a database server and I have had no problems with it
at all. It is about as stable and secure of Linux distro as I have
ever used.

If you do go with them I suggest kicking the CentOS team a few
dollars. They do dang good work.

On Feb 1, 2008 9:16 AM, love U. all <[EMAIL PROTECTED]> wrote:
>
>  i wanna build a production Asterisk box ,will RedHat Linux Enterprise
> Server be more stable than Fedora core Linux  or it makes no significant
> difference
> 
> Express yourself instantly with MSN Messenger! MSN Messenger
>
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[asterisk-users] SIP Softphones and Citrix ?

[d4rk f1br]

Anyone aware of any SIP softphones that might virtualize well with Citrix
presentation server?  I suspect I know the answer already as I have been
researching softphones that work with Cisco CallManager that can be
virtualized if you will with Citrix and have come to learn that its not
something that seems to be doable at this time.  I have to assume that the
issues affecting the virtualization of cisco softphones with Citrix will
come into play with SIP softphones as well.

Seems that the two biggest issues revolve around wrapping the UDP stream up
with the ICA protocol, and possibly issues with the various mics and
speakers and having to interface with them I think.

However, I am also a firm believer that anything is possible, practical well
not usually, and it may just be the time has not come yet for this.  There
is a good article about this over at:

http://www.brianmadden.com/content/article/How-should-Citrix-integrate-VoIP-with-Presentation-Server


Any thoughts, comments or insight into this and your experiences around any
of this is appreciated.
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Re: [asterisk-users] Unicall

[Moises Silva]

Mark, you are confusing terms here.

You DO NOT have Unicall 1.4.9-0.1, libunicall does not even use that
convention for its versions. What you have is astunicall-1.4.9-01.
AstUnicall is just a package with patches and proper versions of
spandsp, libsupertone, libunicall, libmfcr2, zaptel and Asterisk to
run MFC/R2.

Now, having said that, to answer your question, if you just recompile
Asterisk w/o upgrading, then no, you don't need anything. If you want
to upgrade Asterisk, then it depends. Small version upgrades (like
1.4.9 to 1.4.17) probably do not require to change anything, just copy
channels/chan_unicall.c and the Makefile entries of channels/Makefile
and you have good chances of being fine. Upgrading from 1.4 to 1.6
increases the chances of a broken compilation or runtime error. In
general, if you have NO knowledge of C, then you will have to try and
find yourself if it works or not :)

Regards,

Moisés Silva

On Feb 1, 2008 7:24 AM, Mark Welch <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1.
> If we were to update or recompile Asterisk, would we need to do anything
> with Unicall or Zaptel?
>
>
>
> Thanks in advance
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[asterisk-users] Remote Call Center Agents and Asterisk?

[d4rk f1br]

Anyone using Asterisk in a Call Center environment?  And more importantly is
anyone supporting home based remote call center agents with an Asterisk
backend?

My experience with Asterisk is limited, however I have set it up and
installed it previously and had it working for home usage and for simply
playing around.  My background however is with Cisco CallManager, Cisco
IPCCX for call centers as well as a mixed bag of other big name systems.

I am simply researching and investigating different possibilites and
solutions for a project at this point.  Pursuing as many avenues as possible
and trying to setup various test beds and labs if you will to accomplish the
goal of one day rolling out home based remote call center agents.

Look forward to hearing from others about this.  Looking to hear of any
success stories, as well not so successful stories.  Trials and
tribulations, good and bad experiences and where that left you.  I know
others have at the least done exactly what I am doing and have researched
and entertained various ideas regarding this model of home based agents so
hopefully this message can be a catalyst for further disscussion around this
trend.
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Re: [asterisk-users] Problem picking up a call with PickUpChan or PickUp (asterisk-users Digest, Vol 43, Issue 1)

[SIP]

That sounds like a CANCEL message being improperly routed somewhere 
along the line.

Nothing you can do with the config to fix that.

N.



Paul Madley wrote:
> Hi,
>
>   
>> Does anyone of you has a working configuration with SNOM phones that are 
>> able to pickup a call from a flasing LED?
>> 
>
> Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and
> therefore I don't think any config changes will fix it.  We've been told to
> roll back to our previous 1.4.13 installation.  It also seems to manifest
> itself in "ghost ringing" as I've called it; place a call to a SIP
> extension, then put down the handset, yet the destination handset continues
> to ring out.
>
> Sorry I can't be of more use; if anybody else can suggest a workaround it'd
> be greatly appreciated!
>
> Thanks,
>
> P.
>
>
>
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Re: [asterisk-users] Unicall

[Mark Welch]

Thank you Moisés, we are indeed going from 1.4.9 to 1.4.17, we will backup 
channels/chan_unicall.c and the Makefile entries of channels/Makefile and do 
our upgrade to .17.

You are indeed correct on the Unicall, we have astunicall-1.4.9-0.1, I thought 
it was the same thing.  Now I know better :)

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva
Sent: Friday, February 01, 2008 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unicall

Mark, you are confusing terms here.

You DO NOT have Unicall 1.4.9-0.1, libunicall does not even use that
convention for its versions. What you have is astunicall-1.4.9-01.
AstUnicall is just a package with patches and proper versions of
spandsp, libsupertone, libunicall, libmfcr2, zaptel and Asterisk to
run MFC/R2.

Now, having said that, to answer your question, if you just recompile
Asterisk w/o upgrading, then no, you don't need anything. If you want
to upgrade Asterisk, then it depends. Small version upgrades (like
1.4.9 to 1.4.17) probably do not require to change anything, just copy
channels/chan_unicall.c and the Makefile entries of channels/Makefile
and you have good chances of being fine. Upgrading from 1.4 to 1.6
increases the chances of a broken compilation or runtime error. In
general, if you have NO knowledge of C, then you will have to try and
find yourself if it works or not :)

Regards,

Moisés Silva

On Feb 1, 2008 7:24 AM, Mark Welch <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1.
> If we were to update or recompile Asterisk, would we need to do anything
> with Unicall or Zaptel?
>
>
>
> Thanks in advance
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Re: [asterisk-users] G729 version to be downloaded for my machines

[Mindaugas Kezys]

Download for Pentium4

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Wednesday, January 30, 2008 10:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] G729 version to be downloaded for my machines

Hi List;

The output of cat /proc/cpuinfo giving a [Intel (R)
Pentium (R) D] so what is the g729 version I have to
download to work with my machine?

Any help?
Regards
Bilal


  

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Re: [asterisk-users] Meetme voice quality problems

[Matthew J. Roth]

Administrator TOOTAI wrote:
> This is not true if you're using B410P cards. We always face timing 
> problem as we can't -Asterisk stability issues- add X100P or TDM400P 
> with those cards
Daniel,

I thought that using an empty TDM400P as a timing source may no longer 
be the best solution due to the emergence of new stable timing sources 
(such as HPET), but this is an interesting issue.  Are you stating that 
you can't put an X100P or a TDM400P with no lines attached alongside a 
B410P because it impacts the stability of Asterisk?  Do you have any 
idea why?  Can't the B410P be used as a timing source?  What have you 
done to provide stable timing?

I know that's a lot of questions, but I'm genuinely curious.  It seems 
very strange that a TDM400P in timingonly mode and no lines attached 
would have any impact on Asterisk's stability.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


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Re: [asterisk-users] Problem picking up a call with PickUpChan or PickUp (asterisk-users Digest, Vol 43, Issue 1)

[Paul Madley]

Hi,

> Does anyone of you has a working configuration with SNOM phones that are 
> able to pickup a call from a flasing LED?

Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and
therefore I don't think any config changes will fix it.  We've been told to
roll back to our previous 1.4.13 installation.  It also seems to manifest
itself in "ghost ringing" as I've called it; place a call to a SIP
extension, then put down the handset, yet the destination handset continues
to ring out.

Sorry I can't be of more use; if anybody else can suggest a workaround it'd
be greatly appreciated!

Thanks,

P.



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Re: [asterisk-users] Enterprise or Fedora?

[Milton Calnek]

Matthew J. Roth wrote:
> love U.all wrote:
>> i wanna build a production Asterisk box ,will RedHat Linux Enterprise 
>> Server be more stable than Fedora core Linux  or it makes no 
>> significant difference
> I started out running Fedora, but I have migrated away from it for a few 
> reasons.  Fedora has a very short life cycle and since the demise of the 
> legacy project there is no support for releases that are over a year 

I'd take it a step further... if you want asterisk on a stable platform, 
I'd suggest trixbox.  It is CentOS with an additional asterisk repo.

It also comes with several additional apps like SugarCRM and web based 
management.

-- 
Milton Calnek BSc, A/Slt(Ret.)
[EMAIL PROTECTED]
306-717-8737


-- 
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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Mindaugas Kezys]

Hello,

For such cases we usually suggest to put 2 boxes in your infrastructure: 

1. Main billing gateway - where all PBX'es are connected (all client's remote 
PBX'es and your Local PBX)
2. Local PBX - where user's without PBX'es are connected

Then user connects in following way:

User -> Local PBX -> Main GTW -> PSTN

That way you will be save from transfer issue and all your clients will be able 
to transfer their calls on Local PBX.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man
Sent: Wednesday, January 30, 2008 12:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's


> - Original Message 

> From: Matt <[EMAIL PROTECTED]>

> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 

> Sent: Tuesday, 29 January, 2008 9:24:14 PM

> Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's


>

> The provider can disable transfers (which is what we do), but why can a PBX 
> not still allow it?  Our PBX customers all can do 
> transferring... but that's because billing isn't needed THERE.  The billing, 
> if any, is done on our end, or their providers end.   
> This really seems like a very small and moot point that is being blown up.

 

Depends how much it could cost you I guess :). If you're not supporting 
transfers it's a moot point if you are it's a bit more interesting.

> If the receptionist needs to transfer the call, then she should be able to do 
> that within the confines of her PBX... the transfer of
> her call should NEVER go back out her PBX back to the supplier, for if it 
> does, her PBX now loses control of that call.

 

Our customer base is residential and small business. They don't want to either 
pay for or support another a PBX thats what they've come to us for in the first 
place a lot of the time.

Regards,

Greyman.








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Re: [asterisk-users] pulling my hair out over voicemail

[John Von Essen]

Ok, I have made some progress debugging this. I dont believe it has 
anything to do with asterisk or my phone.  Rather I think it is an 
issues with STUN and/or my Linksys router at home.

The phones I am testing all sit behind a NAT'd firewall, your basic 
Linksys router for the Home DSL user.

The phones all of STUN setup, and the STUN server IP is the IP of the 
asterisk server - which is purely public.

I was able to duplicate the problem with not being able to hear the 
voicemail greeting by doing the following:

Turn off all the phones, and power cycle my Linksys, then turn on 1 
phone. That one phone will then work, and you can hear voicemail 
greeting.

The I turn on the second phone. Then voicemail greeting breaks, and you 
cant hear it when you dial into voicemail. If I unplug the first phone, 
and power cycle the Linksys again, the second phone will begin to work.

So the question is, does this behavior make sense?

I assumed with an STUN server I could have multiple phones behind my 
Linksys firewall, now it appears I can only have one. Is it a Linksys 
bug, or a general known issue? Do I need to run multiple STUN servers?

Thanks
John



On Jan 31, 2008, at 1:00 PM, Shane D wrote:

> Very odd. Could you try taking the mailbox line out of sip.conf and
> see what happens?
>
> On 1/31/08, John Von Essen <[EMAIL PROTECTED]> wrote:
>> Here are my configs:
>>
>>
>> sip.conf:
>>
>> [general]
>> context=default
>> bindport=5060
>> bindaddr=0.0.0.0
>> disallow=all
>> allow=ulaw
>>
>> [6000]
>> type=friend
>> secret=letmein
>> host=dynamic
>> dtmfmode=rfc2833
>> mailbox=6000
>> context=default
>>
>> extensions.conf:
>>
>> [default]
>> exten => 1000,1,Ringing
>> exten => 1000,2,Wait(2)
>> exten => 1000,3,VoicemailMain
>>
>> Calling from phone to phone is fine, and inbound and outbound calling
>> is fine. But when I call voicemail, I dont hear anything.
>>
>> When I view console in CLI I see this when attempting to dial the
>> voicemail extension:
>>
>>  -- Executing [EMAIL PROTECTED]:1] Ringing("SIP/6001-081d65c8", "") in
>> new stack
>>  -- Executing [EMAIL PROTECTED]:2] Wait("SIP/6001-081d65c8", "2") in 
>> new
>> stack
>>  -- Executing [EMAIL PROTECTED]:3] VoiceMailMain("SIP/6001-081d65c8",
>> "[EMAIL PROTECTED]") in new stack
>>  --  Playing 'vm-login' (language 'en')
>> [Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate:
>> Couldn't read username
>> Really destroying SIP dialog '[EMAIL PROTECTED]' Method:
>> BYE
>>
>> So it plays the greetings, and is working, I just cant hear it.
>>
>> -john
>>
>>
>>
>>
>>
>> On Jan 31, 2008, at 3:00 AM, Andrew Joakimsen wrote:
>>
>>> On Jan 31, 2008 12:30 AM, John Von Essen <[EMAIL PROTECTED]> wrote:

 Any ideas what could be going on? I tried tweaking the extension 
 1000
 so it looks like:
>>>
>>> Maybe the SIP  config is wrong?
>>>

 Where 6000 is my mailbox. But still nothing, when I dial 1000, it 
 just
 goes silent.
>>>
>>> Can you places other calls from that new phone?
>>>
 Please help. This is driving me nuts. I even tried re-installing
 asterisk from scratch - no change.
>>>
>>> What version?
>>>
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>>
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>
>
> -- 
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
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[asterisk-users] Codec preference selection, codec negotiation

[bilal ghayyad]

Hi List;

Is it possible to do configuration at the user context
to let him use codec1 if destination support and if
not, then use codec 2?

For example, to let user1 use codec g729 if he needs
to call user2 and user2 support g729, and if user2
does not support g729 then use g711 (alaw or ulaw), is
it possible? I think this kind of settings required
codec negotiation and I do not know if Asterisk has
such capability.

Any help?
Regards
Bilal


  

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Re: [asterisk-users] Enterprise or Fedora?

[Mindaugas Kezys]

My suggestion - use the distro which you know best.

We use Debian (200+ installations). It works stable for us because we know how 
to achieve it.

Others use Fedora/Centos - because they are experts in these systems.

Stability and performance of the system does not depend on the distro - only on 
person who built this system.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LWATCDR
Sent: Friday, February 01, 2008 6:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Enterprise or Fedora?

We are using Fedora because that is what the company we got our system
from recommended.
If I was doing the system myself I would throw in my vote for CentOS.
I am using it for a database server and I have had no problems with it
at all. It is about as stable and secure of Linux distro as I have
ever used.

If you do go with them I suggest kicking the CentOS team a few
dollars. They do dang good work.

On Feb 1, 2008 9:16 AM, love U. all <[EMAIL PROTECTED]> wrote:
>
>  i wanna build a production Asterisk box ,will RedHat Linux Enterprise
> Server be more stable than Fedora core Linux  or it makes no significant
> difference
> 
> Express yourself instantly with MSN Messenger! MSN Messenger
>
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Re: [asterisk-users] Remote Call Center Agents and Asterisk?

[Steve Totaro]

On Feb 1, 2008 10:33 AM, d4rk f1br <[EMAIL PROTECTED]> wrote:
> Anyone using Asterisk in a Call Center environment?  And more importantly is
> anyone supporting home based remote call center agents with an Asterisk
> backend?
>
> My experience with Asterisk is limited, however I have set it up and
> installed it previously and had it working for home usage and for simply
> playing around.  My background however is with Cisco CallManager, Cisco
> IPCCX for call centers as well as a mixed bag of other big name systems.
>
> I am simply researching and investigating different possibilites and
> solutions for a project at this point.  Pursuing as many avenues as possible
> and trying to setup various test beds and labs if you will to accomplish the
> goal of one day rolling out home based remote call center agents.
>
> Look forward to hearing from others about this.  Looking to hear of any
> success stories, as well not so successful stories.  Trials and
> tribulations, good and bad experiences and where that left you.  I know
> others have at the least done exactly what I am doing and have researched
> and entertained various ideas regarding this model of home based agents so
> hopefully this message can be a catalyst for further disscussion around this
> trend.


I have had several very successful implementations.  Some small
~50-100 agents, and some larger, around 500 agents.

The trick is keeping the agents honest since there is no supervisor
standing behind them.  You want to establish a minimum standard for
the home agent.

Recording of calls for sound quality and agent evaluation will be
critical for QA as well as ChanSpy and "whisper coaching" which I
believe is available in 1.4 (?)

Use AJAX and Jabber ActiveX controls to control your CRM (web based of course).

You could even ship them a "Kit" which contains a router running
DD-WRT or OpenWRT setup with an OpenVPN tunnel back to your data
center, an ATA, and a headset.

There are several benefits to sending the router.  Obviously, VPN.
Then you also have something to SSH into and do testing like ping,
traceroute, test throughput.  You could even create some kind of app
(if it doesn't already exist) to regularly run these diagnostics and
upload them to you.

That is just a few ideas but I think the main thing you will run into
is agents not logging off or somehow trying to beat the system.  That
is what I see time and time again.

Thanks,
Steve Totaro

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Re: [asterisk-users] Remote Call Center Agents and Asterisk?

[Steve Totaro]

On Feb 1, 2008 10:33 AM, d4rk f1br <[EMAIL PROTECTED]> wrote:
> Anyone using Asterisk in a Call Center environment?  And more importantly is
> anyone supporting home based remote call center agents with an Asterisk
> backend?
>
> My experience with Asterisk is limited, however I have set it up and
> installed it previously and had it working for home usage and for simply
> playing around.  My background however is with Cisco CallManager, Cisco
> IPCCX for call centers as well as a mixed bag of other big name systems.
>
> I am simply researching and investigating different possibilites and
> solutions for a project at this point.  Pursuing as many avenues as possible
> and trying to setup various test beds and labs if you will to accomplish the
> goal of one day rolling out home based remote call center agents.
>
> Look forward to hearing from others about this.  Looking to hear of any
> success stories, as well not so successful stories.  Trials and
> tribulations, good and bad experiences and where that left you.  I know
> others have at the least done exactly what I am doing and have researched
> and entertained various ideas regarding this model of home based agents so
> hopefully this message can be a catalyst for further disscussion around this
> trend.


I have had several very successful implementations.  Some small
~50-100 agents, and some larger, around 500 agents.

The trick is keeping the agents honest since there is no supervisor
standing behind them.  You want to establish a minimum standard for
the home agent.

Recording of calls for sound quality and agent evaluation will be
critical for QA as well as ChanSpy and "whisper coaching" which I
believe is available in 1.4 (?)

Use AJAX and Jabber ActiveX controls to control your CRM (web based of course).

You could even ship them a "Kit" which contains a router running
DD-WRT or OpenWRT setup with an OpenVPN tunnel back to your data
center, an ATA, and a headset.

There are several benefits to sending the router.  Obviously, VPN.
Then you also have something to SSH into and do testing like ping,
traceroute, test throughput.  You could even create some kind of app
(if it doesn't already exist) to regularly run these diagnostics and
upload them to you.

That is just a few ideas but I think the main thing you will run into
is agents not logging off or somehow trying to beat the system.  That
is what I see time and time again.

Thanks,
Steve Totaro

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Re: [asterisk-users] Remote Call Center Agents and Asterisk?

[EdPimentl]

Hello Steve,

You are right on track and this is also what we have done with pretty good
results.
Of course now with Flex/Air there are a number of ways to enhance the
service for the
Customer/Agent

Ed

Mail:   edpimentl[at]gmail.com
Voip:   edpimentl [SKype | GoogleTalk ]

http://agileoss.com (Web2.0 and SOA Development )
http://mobiquity.ws (Private Label Social Network)
http://youbiquity.ws (Power of One for all Social Networks)
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Re: [asterisk-users] CallerID shows wrong values in manager interface

[Mojo with Horan & Company, LLC]

Trust me, I don't WANT you to look at my code, it's butt-ugly! lol, lust 
kidding...  -- but at http://www.astsee.com/ you can download the source 
code to my AstSee project -- it may provide some insight into what needs 
to be (or CAN be) gleaned from asterisk.   I struggled with all this a 
year or more ago and manged to get it to work fairly as expected.  A 
problem you might notice is that I believe I mistakenly update my 
internal arrays before asterisk's manager interface has sent a complete 
packet... argh   But you can see me dealing with NewState, NewExten, 
NewChannel, etc etc and what I do with them :)

Moj

Devraj Mukherjee wrote:
> Thanks all :)
>
> Appreciate it.
>
> On Feb 1, 2008 12:04 PM, Ex Vito <[EMAIL PROTECTED]> wrote:
>   
>>   I've struggled with this recently. In short:
>>
>>
>>   - Observed behaviour is expected as of asterisk 1.2 and later,
>> as previously described by Mojo
>>
>>   - If you want to get the caller id for the channel calling (dialling)
>> into that channel for that specific Newstate: Ringing event, you
>> can use the 'o' flag to the Dial command; in this case you'll get
>> old asterisk 1.0 behaviour -- do you really want to depend on
>> such an old behaviour ? well I decided I didn't...
>>
>>   - Otherwise, you'll need to track other events (IIRC, at least, Dial,
>> AgentCalled, Newstate, etc) in the AMI so as to know who is calling
>> who at a given instant
>>
>>   - BEWARE: if memory serves me right (search the list archives in the 
>> Nov/Dec
>> timeframe), the behaviour is not 100% homogeneous for different channel
>> types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from
>> one channel to the other is that a) at times you get the Dial
>> event first then the
>> Newstate: Ringing event; and that b) with other/different
>> orig/dest channel types
>> you'll get the events in the reverse order... Nothing much but: i)
>> you'll have to
>> track them either way and ii) it reveals that the AMI events
>> aren't 100% clean!!!
>>
>>   :/
>> --
>>   exvito
>>
>>
>> On Feb 1, 2008 12:08 AM, Mojo with Horan & Company, LLC
>> <[EMAIL PROTECTED]> wrote:
>> 
>>> The snippet is asterisk telling you "I'm just letting you know that the
>>> correct caller id for Channel: SIP/103-098500d8 is CallerID: 103"
>>>
>>> This is absolutely correct, it's just not a piece of information you
>>> expected to be receiving at that point.
>>>
>>> You probably also received a packet like that with the following:
>>>  Channel: SIP/101-
>>>  CallerID: 101
>>> telling you, again, the caller id for only that channel.
>>>
>>> Moj
>>>
>>>   
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>
>
>
>   


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Re: [asterisk-users] Enterprise or Fedora?

[Steve Edwards]

On Fri, 1 Feb 2008, Milton Calnek wrote:

>
>
> Matthew J. Roth wrote:
>> love U.all wrote:
>>> i wanna build a production Asterisk box ,will RedHat Linux Enterprise
>>> Server be more stable than Fedora core Linux  or it makes no
>>> significant difference
>> I started out running Fedora, but I have migrated away from it for a few
>> reasons.  Fedora has a very short life cycle and since the demise of the
>> legacy project there is no support for releases that are over a year
>
> I'd take it a step further... if you want asterisk on a stable platform,
> I'd suggest trixbox.  It is CentOS with an additional asterisk repo.
>
> It also comes with several additional apps like SugarCRM and web based
> management.

But all that cruft is not always appropriate -- parts left out don't get 
broke. They also don't need to be updated, suck up memory, cpu, disk or 
get in the way of learning how Asterisk really works.

I like to start with a minimal CentOS install -- de-select everything and 
then "yum install" just the packages you actually need.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] Astersik Transcoder support

[Charles Feng]

Hello All:

Does the Asterisk support to insert an off the board transcoder for a call?

Thanks,

Charles


  

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Re: [asterisk-users] Astersik Transcoder support

[Simon Elliston Ball]

http://www.digium.com/en/products/voice/tc400b.php


Simon Elliston Ball
[EMAIL PROTECTED]



On 1 Feb 2008, at 17:29, Charles Feng wrote:

> Hello All:
>
> Does the Asterisk support to insert an off the board transcoder  
> for a call?
>
> Thanks,
>
> Charles
>
> Never miss a thing. Make Yahoo your homepage.  
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Re: [asterisk-users] Enterprise or Fedora?

[Tzafrir Cohen]

On Fri, Feb 01, 2008 at 09:58:28AM -0600, Milton Calnek wrote:
> 
> 
> Matthew J. Roth wrote:
> > love U.all wrote:
> >> i wanna build a production Asterisk box ,will RedHat Linux Enterprise 
> >> Server be more stable than Fedora core Linux  or it makes no 
> >> significant difference
> > I started out running Fedora, but I have migrated away from it for a few 
> > reasons.  Fedora has a very short life cycle and since the demise of the 
> > legacy project there is no support for releases that are over a year 
> 
> I'd take it a step further... if you want asterisk on a stable platform, 
> I'd suggest trixbox.  It is CentOS with an additional asterisk repo.
> 
> It also comes with several additional apps like SugarCRM and web based 
> management.

I'm sorry, but Trixbox (CE) and its like are modified versions of
CentOS, and those modifications are not always so standard:

1. Many changes in /usr/local and such rather than in packages. 
2. Kernel never gets updated. Even latest trixbox 2.2 versions are
   shipped with kernel from CentOS 4.3. CentOS has issued many updates
   since. Try to build your own kernel stuff on it.
3. It uses FreePBX, and this has its pros and cons. It is surely a
   comlpicated system to start with. If you happen not to like it,
   you're heading for much work in removing it.

So it's not simply "CentOS".

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Asterisk 1.4.17 and Teliax DTMF

[Carlos Chavez]

I am having a problem with DTMF when sending calls through Teliax
(SIP).  In the peer for teliax I defined dtmfmode=rfc2833 and for the
most part it is working.  The problem always happens when a user is
trying to call a conference system.  They simply cannot get into the
conference because DTMF is not understood.  If I dial from a land line I
can get in with no problems.

Any tweaks recommended for DTMF and Teliax?

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Enterprise or Fedora?

[shadowym]

I cannot think of a single reason to use Fedora for a production anything
when there are alternatives like CentOS.  Fedora is bleeding edge stuff and
constantly changing.

-Original Message-
From: Matthew J. Roth [mailto:[EMAIL PROTECTED] 
Sent: Friday, February 01, 2008 7:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Enterprise or Fedora?

love U.all wrote:
> i wanna build a production Asterisk box ,will RedHat Linux Enterprise 
> Server be more stable than Fedora core Linux  or it makes no 
> significant difference
I started out running Fedora, but I have migrated away from it for a few 
reasons.  Fedora has a very short life cycle and since the demise of the 
legacy project there is no support for releases that are over a year 
old.  It is also considered a bleeding-edge distribution and is a 
testing ground for new applications and features.  As such, it has 
frequent updates (including the kernel) that aren't guaranteed not to 
break your system.

If stability and long-term support are your goals, I recommend taking a 
look at CentOS .  It is a binary-compatible 
clone of Red Hat Enterprise Linux that's free and has a very long 
support period.  It's basically RHEL without the paid support contract.  
My migration to CentOS was painless, because the file system and 
configuration are practically identical to those of Fedora.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer






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[asterisk-users] Bypassing a Auth on Invite or Forbiden?

[Bryan Cramer]

Hello,

I have 2 asterisk servers that are not working well together.  One is  
acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX  
devices.  And the other is acting like my sip gateway (PBX02) to  
various providers.  They are both on a private network and should be  
trusting each others IP 100%.  But the PBX02 challenges PBX01's  
requests all the time even though insecure=invite is set.   Here are  
the stanzas for each server:


PBX01

[pbx01topbx02]
type=friend
context=incomingDefault
host=10.10.10.2
qualify=600
disallow=all
allow=ulaw
canreinvite=no
insecure=invite
accountcode=pbx02
;TESTS;
;[EMAIL PROTECTED]
;fromuser=pbx01topbx02
;username=pbx01topbx02
;secret=j48dj7rjd9023jd
sendrpid=yes
trustrpid=yes


PBX02

[pbx01topbx02]
type=friend
host=10.10.10.3
qualify=600
context=dialOutPatternsAll
disallow=all
allow=ulaw
canreinvite=no
insecure=invite
accountcode=pbx01
;TESTS;
;[EMAIL PROTECTED]
;username=pbx01topbx02
;fromuser=pbx01topbx02
;secret=j48dj7rjd9023jd
sendrpid=yes
trustrpid=yes

Layout

SIPDEV -REGISTERD-> PBX01 -> PBX02 -> PROVIDER

With the above settings above I should be allowed to send calls as I  
like through the 2 boxes, but I can't I get the following messages:


PBX01
-- Executing [EMAIL PROTECTED]:7] Dial("IAX2/4161231234-2",  
"SIP/pbx01topbx02/16041231234") in new stack

-- Called pbx01topbx02/16041231234
[Feb  1 11:35:18] NOTICE[13507]: chan_sip.c:11983  
handle_response_invite: Failed to authenticate on INVITE to  
'"4161231234" ;tag=as4a9e0ae9'

-- SIP/pbx01topbx02-009c31d0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

Then on PBX02 I don't get any debug/verbose messages unless I do a  
sip debug then I get:


<--- SIP read from 10.10.10.3:5060 --->
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK349431bd;rport
From: "4161231234" ;tag=as5c5aac02
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "4161231234" [EMAIL PROTECTED]>;privacy=off;screen=yes

Date: Fri, 01 Feb 2008 19:40:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 13479 13479 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 11026 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<->
--- (15 headers 12 lines) ---
Sending to 10.10.10.3 : 5060 (NAT)
Using INVITE request as basis request -  
[EMAIL PROTECTED]


<--- Reliably Transmitting (no NAT) to 10.10.10.3:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP  
10.10.10.3:5060;branch=z9hG4bK349431bd;received=10.10.10.3;rport=5060

From: "4161231234" ;tag=as5c5aac02
To: ;tag=as19dfe789
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",  
nonce="6d7284c8"

Content-Length: 0

Any insights as to why insecure invite is not taking effect?

Bryan
_
Bryan Cramer

PO Box 616
Sechelt BC, V0N 3A0
Web: www.yellowions.com
Tel: 604-773-4580

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Re: [asterisk-users] LDAP support

[Gavin Henry]

There a realtime LDAP driver now in 1.6beta2

On 23/01/2008, Cavalera Claudio Luigi <[EMAIL PROTECTED]> wrote:
> Hello,
> I've found this information about asterisk and LDAP:
> http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
> which can be out of date.
>
> I'm trying this http://www.mezzo.net/asterisk/app_ldap.html
> however I'm facing the same problems as this unanswered:
> http://forums.digium.com/viewtopic.php?p=42591&sid=05e1d00ab6f9848f4e7b6
> 39d66cc6d79
> Does anybody know how to solve this issue?
>
> Moreover I would like to understand if someone is using LDAP (for
> iax.conf) and with which asterisk "plugin" (e.g. app_ldap,
> Asterisk::LDAP Perl module, etc..).
>
> Best Regards,
> Claudio
>
>
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http://www.suretecsystems.com/services/openldap/

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Re: [asterisk-users] Enterprise or Fedora?

[Milton Calnek]

Matthew J. Roth wrote:
> love U.all wrote:
>> i wanna build a production Asterisk box ,will RedHat Linux Enterprise 
>> Server be more stable than Fedora core Linux  or it makes no 
>> significant difference
> I started out running Fedora, but I have migrated away from it for a few 
> reasons.  Fedora has a very short life cycle and since the demise of the 
> legacy project there is no support for releases that are over a year 
> old.  It is also considered a bleeding-edge distribution and is a 
> testing ground for new applications and features.  As such, it has 
> frequent updates (including the kernel) that aren't guaranteed not to 
> break your system.
> 
> If stability and long-term support are your goals, I recommend taking a 
> look at CentOS .  It is a binary-compatible 
> clone of Red Hat Enterprise Linux that's free and has a very long 
> support period.  It's basically RHEL without the paid support contract.  
> My migration to CentOS was painless, because the file system and 
> configuration are practically identical to those of Fedora.
> 
> Regards,
> 
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
> 
> 
> 
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-- 
Milton Calnek BSc, A/Slt(Ret.)
[EMAIL PROTECTED]
306-717-8737



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[asterisk-users] It's about time! -- Digium PCI-Express Cards

[Matt]

Just noticed this today:

Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based Echo
Cancellation Module

It's about time Digium got on the ball and made PCI-e cards.   What are
people's experiences with this card?  Anyone know if there are plans for a
PCI-e analog card for FXO use?
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[asterisk-users] QueueMember event/LastCall Variable - Format?

[Lee Jenkins]

Hi all,

What format is the LastCall variable of QueueMember event?  I'm looking at: 
1201897536 for instance.

Thanks !

-- 
Warm Regards,

Lee

"Everything I needed to learn in life, I learned selling encyclopedias door to 
door."

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Re: [asterisk-users] QueueMember event/LastCall Variable - Format?

[Jared Smith]

On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote:
> What format is the LastCall variable of QueueMember event?  I'm looking at: 
> 1201897536 for instance.

Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I
recall.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards

[Jason Parker]

Matt wrote:
> Just noticed this today:
> 
> Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based
> Echo Cancellation Module
> 
> 
> It's about time Digium got on the ball and made PCI-e cards.   What are
> people's experiences with this card?  Anyone know if there are plans for
> a PCI-e analog card for FXO use?
> 

Digium already makes PCI Express analog cards - AEX800 and AEX2400.

-- 
Jason Parker
Digium

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Re: [asterisk-users] SIP Softphones and Citrix ?

[Tim H. Panton]

There is an option you might consider (if you are starting from scratch).

Don't use citrix. Write a web app.

Then embed a softphone in that web app.

Tim.


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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Grey Man]

- Original Message 
> From: Mindaugas Kezys <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Friday, 1 February, 2008 4:04:30 PM
> Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
> 
>For 
such 
cases 
we 
usually 
suggest 
to 
put 
2 
boxes 
in 
your 
infrastructure: 
>
> 1. 
Main 
billing 
gateway 
- 
where 
all 
PBX'es 
are 
connected 
(all 
client's 
remote 
PBX'es 
and 
your 
Local 
PBX)
> 2. 
Local 
PBX 
- 
where 
user's 
without 
PBX'es 
are 
connected
> 
> Then 
user 
connects 
in 
following 
way:
> 
> User 
-> 
Local 
PBX 
-> 
Main 
GTW 
-> 
PSTN
>
> That 
way 
you 
will 
be 
save 
from 
transfer 
issue 
and 
all 
your 
clients 
will 
be 
able 
to 
transfer 
their 
calls 
on 
Local 
PBX.

Hi Mindaugas,

That's a good tip, thanks for that.

My concerns would be that the call path is now running through two Asterisk 
servers and that could add some quality problems, probably negligible though. 
The other concern would be that for fault tolerance we'd now need double the 
number of servers. If we currently require 3 Asterisk load balanced servers 
then now we are going to need 6. It's an idea worth toying around with though.

Maybe we could specify that all customers that required the ability to transfer 
had to use server x and then from the SIP Proxy only allow REFER requests to 
that server.

Regards,

Greyman.



  Get the name you always wanted with the new y7mail email address.
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Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards

[Matt]

Jason.. great!   How long have those analog cards been out?  I don't see
them on my suppliers list.


>
> Digium already makes PCI Express analog cards - AEX800 and AEX2400.
>
> --
> Jason Parker
> Digium
>
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Re: [asterisk-users] SIP Softphones and Citrix ?

[Zoa]

I'm working for zoiper.com and i'm willing to help out with ours when 
needed.

Zoa


d4rk f1br wrote:
> Anyone aware of any SIP softphones that might virtualize well with 
> Citrix presentation server?  I suspect I know the answer already as I 
> have been researching softphones that work with Cisco CallManager that 
> can be virtualized if you will with Citrix and have come to learn that 
> its not something that seems to be doable at this time.  I have to 
> assume that the issues affecting the virtualization of cisco 
> softphones with Citrix will come into play with SIP softphones as well.
>  
> Seems that the two biggest issues revolve around wrapping the UDP 
> stream up with the ICA protocol, and possibly issues with the various 
> mics and speakers and having to interface with them I think.
>  
> However, I am also a firm believer that anything is possible, 
> practical well not usually, and it may just be the time has not come 
> yet for this.  There is a good article about this over at:
>  
> http://www.brianmadden.com/content/article/How-should-Citrix-integrate-VoIP-with-Presentation-Server
>  
>  
> Any thoughts, comments or insight into this and your experiences 
> around any of this is appreciated.
> 
>
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Re: [asterisk-users] QueueMember event/LastCall Variable - Format?

[Lee Jenkins]

Jared Smith wrote:
> On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote:
>> What format is the LastCall variable of QueueMember event?  I'm looking at: 
>> 1201897536 for instance.
> 
> Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I
> recall.
> 

Thanks.
-- 
Warm Regards,

Lee

"Everything I needed to learn in life, I learned selling encyclopedias door to 
door."

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Re: [asterisk-users] Asterisk 1.4.17 and Teliax DTMF

[Ira]

At 10:57 AM 2/1/2008, you wrote:
> Any tweaks recommended for DTMF and Teliax?

try:

dtmfmode=auto

That's what mine is now after rfc2833 stopped working.

Ira 


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Re: [asterisk-users] Enterprise or Fedora?

[Benny Amorsen]

"shadowym" <[EMAIL PROTECTED]> writes:

> I cannot think of a single reason to use Fedora for a production anything
> when there are alternatives like CentOS.  Fedora is bleeding edge stuff and
> constantly changing.

The advantage of Fedora is that it is very actively maintained -- and
asterisk is only a yum install asterisk away!


/Benny



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Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards

[Ron Joffe]

On Friday 01 February 2008 15:31, Matt wrote:
> It's about time Digium got on the ball and made PCI-e cards.   What are
> people's experiences with this card?  Anyone know if there are plans for a
> PCI-e analog card for FXO use?

I have been using 220B's for about 6 months. I have about 20 of them out in 
the field. I have not had any issues with them, and feedback is positive.

Ron


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[asterisk-users] Trying to make SIP calls through Asterisk with anonymous connection

[Royce Souther]

I am trying to setup SIP to SIP calling between Asterisk managed networks. I
want to make it so that people can call SIP:[EMAIL PROTECTED] and they
connect to my Asterisk and get my external IVR then they can dial my
extension or navigate extensions just like they would if they had called
using a PSTN line. I also want to call other people using my Asterisk and
dialing an external SIP like so SIP:[EMAIL PROTECTED] I want out going SIP
calls to me managed by my Asterisk so I can transfer them to other people in
my office or conference or use any of the other great features that Asterisk
provides. I do not want to go SIP direct to SIP, I want to go SIP to
Asterisk to Asterisk to SIP and connect to the far end Asterisk without
requiring me to register my Asterisk server with the far end Asterisk
server.

For testing I have setup two servers running Asterisk. Both are on the
Internet with static IP addresses and behind firewalls. The firewalls are
configured to allow TCP & UDP ports 5060 to 5082 and 10001 to 2 to
connect directly to the Asterisk servers. This allows SIP and RTP
connections from the outside. I have tested with Twinkle (a Linux softphone)
and can connect to a registered account with NAT from external IPs. I have
also set the Asterisk servers to allow incoming anonymous SIP calls to
connect to the from-external.  When I try to dial SIP:[EMAIL PROTECTED]
Asterisk tries to dial the some_extension on my local network not the other
network. I reconnect to the running asterisk using -r and watch when I dial
and it does not report the @other_url only the some_extension.

I am not having much luck finding the documentation I need. Can someone
point me to a How-To on doing this?

-- 
Open Source: To innovate then create
Proprietary: To imitate then litigate
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[asterisk-users] X-Lite Softphone keeps de-registering?

[Doug]

The client is travelling much of the time.

Is there some way that he can use Port 80 so
that the firewalls that he is behind won't
block the connection?

Any other hints or suggestions are very
welcome!


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[asterisk-users] "Real" API for Perl?

[Ken D'Ambrosio]

Hi, all.  I've used the perl/AGI interface, and... well, I found it kind
of hokey.  Granted, this was in 1.2 days -- perhaps things have changed. 
Regardless, I guess I have two questions:
1) Has the Perl/AGI "binding" improved since then?
2) Is there any chance of a "real" API for Perl?

Thanks much!

-Ken




-- 
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Re: [asterisk-users] Asterisk 1.4.17 and Teliax DTMF

[Darrick Hartman (lists)]

Carlos Chavez wrote:
>   I am having a problem with DTMF when sending calls through Teliax
> (SIP).  In the peer for teliax I defined dtmfmode=rfc2833 and for the
> most part it is working.  The problem always happens when a user is
> trying to call a conference system.  They simply cannot get into the
> conference because DTMF is not understood.  If I dial from a land line I
> can get in with no problems.
> 
>   Any tweaks recommended for DTMF and Teliax?

I've had no issues with our Teliax accounts since switching to 1.4.x.  I 
stayed back on 1.4.16.2 so far because of one issue with parking calls.
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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[asterisk-users] IAX Registraion Refresh

[Douglas Garstang]

I have Asterisk 1.4 registering via IAX to another Asterisk machine.
How can I change the default registration timeout of 60s?
I need my Asterisk box to register every HOUR Anyone?

Editting source isn't an option.

Doug.





  

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[asterisk-users] Echo() app doesn't work

[Yassen Damyanov]

Hello list,

New to asterisk and to the list (although experienced in Unix/Linux
administration).

Short problem description:
--
I cannot get the Echo() application to run on any 32bit platform I can get my
hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have
runs just fine. In all cases asterisk log shows the same -- that Echo() is
executed

Details:

A. Platforms:

-- AsteriskNOW 0.6 beta 32bit, updated;

-- Debian Etch 32bit with stock kernel and native debian-packaged asterisk 1.2

-- Debian Etch 32bit with custom compiled kernel ("Timer frequency" 1000 Hz and
couple more tweaks) and latest stable asterisk (1.4.17) compiled from source

-- Ubuntiu 7.10 Server with native ubuntu packages (1.4.10)

-- xUbuntu 7.10 Desktop on x86_64 (=amd64) with native ubuntu packages (1.4.10)

Echo() works only on the 64-bit setup. Does not work for all other cases.

The Playback() app works fine in *all* cases.

(The microphone is tested and works fine, so it's not that simple!)

For some of the setups I established two separate extensions and they could
talk to each other (so important things work, yes).

The logs show the same, that is, just what would be normal:

-cut here---
Asterisk Ready.
*CLI> -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569
-- Accepting UNAUTHENTICATED call from 192.168.2.3:
   > requested format = gsm,
   > requested prefs = (),
   > actual format = gsm,
   > host prefs = (),
   > priority = mine
-- Executing [EMAIL PROTECTED]:1] Verbose("IAX2/yassen-2", "1|Echo test
application") in new stack
 Echo test application
-- Executing [EMAIL PROTECTED]:2] Echo("IAX2/yassen-2", "") in new stack
  == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2'
-- Hungup 'IAX2/yassen-2'
-cut here---

My extentions.conf:
-cut here---
[globals]

[general]

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[outgoing_calls]

[incoming_calls]

[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()

exten => 501,1,Verbose(1|Playback test application)
exten => 501,n,Playback(vm-review)
exten => 501,n,Wait(1)
exten => 501,n,Hangup()

[phones]
include => internal
-cut here---

My iax.conf:
-cut here---
[general]
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
autokill=yes

[yassen]
type=friend
host=dynamic
context=phones
-cut here---

Anyone having a suggestion what might be the reason for the nonworking Echo() ?
 I am really stuck; Google could not really help. Any ideas would be highly
appreciated!

Thanks in advance,
Yassen

--

Yassen Damyanov
Adelie Ltd.


  

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[asterisk-users] Polycom - Buddy Watch not a choice when adding Speed Dial

[Thermal Wetland]

Hello,

On our Polycom phones we can not activate the Buddy Watch feature.

When you add or edit a contact, the list ends at "Auto Divert".I know it
is the end of the list b/c the down arrow on the right side of the screen
disappears when I get to Auto Divert.

When I add 1 manually to the speed dial file it doesn't change
anything.

The buttons work well for a speed dial.

The icon next the speed dial is 10 dots, in the shape of a keypad.

Anyone else experience this?

Thanks,
Thermal
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Re: [asterisk-users] Polycom - Buddy Watch not a choice when adding Speed Dial

[Matt Darnell]

On Feb 1, 2008 3:53 PM, Thermal Wetland <[EMAIL PROTECTED]> wrote:

> Hello,
>
> On our Polycom phones we can not activate the Buddy Watch feature.
>
> When you add or edit a contact, the list ends at "Auto Divert".I know
> it is the end of the list b/c the down arrow on the right side of the screen
> disappears when I get to Auto Divert.
>
> When I add 1 manually to the speed dial file it doesn't change
> anything.
>
> The buttons work well for a speed dial.
>
> The icon next the speed dial is 10 dots, in the shape of a keypad.
>
> Anyone else experience this?
>
> Thanks,
> Thermal
>


Check your sip.cfg for the line:
feature.1.name="presence" feature.1.enabled="1"

I would imagine that you have enabled="0"

-Matt
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Re: [asterisk-users] Echo() app doesn't work

[Tzafrir Cohen]

On Fri, Feb 01, 2008 at 05:01:56PM -0800, Yassen Damyanov wrote:
> Hello list,
> 
> New to asterisk and to the list (although experienced in Unix/Linux
> administration).
> 
> Short problem description:
> --
> I cannot get the Echo() application to run on any 32bit platform I can get my
> hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have
> runs just fine. In all cases asterisk log shows the same -- that Echo() is
> executed
> 
> Details:
> 
> A. Platforms:
> 
> -- AsteriskNOW 0.6 beta 32bit, updated;
> 
> -- Debian Etch 32bit with stock kernel and native debian-packaged asterisk 1.2
> 
> -- Debian Etch 32bit with custom compiled kernel ("Timer frequency" 1000 Hz 
> and
> couple more tweaks) and latest stable asterisk (1.4.17) compiled from source
> 
> -- Ubuntiu 7.10 Server with native ubuntu packages (1.4.10)
> 
> -- xUbuntu 7.10 Desktop on x86_64 (=amd64) with native ubuntu packages 
> (1.4.10)
> 
> Echo() works only on the 64-bit setup. Does not work for all other cases.
> 
> The Playback() app works fine in *all* cases.
> 
> (The microphone is tested and works fine, so it's not that simple!)
> 
> For some of the setups I established two separate extensions and they could
> talk to each other (so important things work, yes).
> 
> The logs show the same, that is, just what would be normal:
> 
> -cut here---
> Asterisk Ready.
> *CLI> -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569
> -- Accepting UNAUTHENTICATED call from 192.168.2.3:
>> requested format = gsm,
>> requested prefs = (),
>> actual format = gsm,
>> host prefs = (),
>> priority = mine
> -- Executing [EMAIL PROTECTED]:1] Verbose("IAX2/yassen-2", "1|Echo test
> application") in new stack
>  Echo test application
> -- Executing [EMAIL PROTECTED]:2] Echo("IAX2/yassen-2", "") in new stack
>   == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2'
> -- Hungup 'IAX2/yassen-2'
> -cut here---

On which platform is that? Echo is executed, and exists without an
error.

> 
> My extentions.conf:
> -cut here---
> [globals]
> 
> [general]
> 
> [default]
> exten => s,1,Verbose(1|Unrouted call handler)
> exten => s,n,Answer()
> exten => s,n,Wait(1)
> exten => s,n,Playback(tt-weasels)
> exten => s,n,Hangup()
> 
> [outgoing_calls]
> 
> [incoming_calls]
> 
> [internal]
> exten => 500,1,Verbose(1|Echo test application)
> exten => 500,n,Echo()
> exten => 500,n,Hangup()
> 
> exten => 501,1,Verbose(1|Playback test application)
> exten => 501,n,Playback(vm-review)
> exten => 501,n,Wait(1)
> exten => 501,n,Hangup()
> 
> [phones]
> include => internal
> -cut here---
> 
> My iax.conf:
> -cut here---
> [general]
> bandwidth=low
> disallow=lpc10
> jitterbuffer=no
> forcejitterbuffer=no
> autokill=yes
> 
> [yassen]
> type=friend
> host=dynamic
> context=phones
> -cut here---
> 
> Anyone having a suggestion what might be the reason for the nonworking Echo() 
> ?
>  I am really stuck; Google could not really help. Any ideas would be highly
> appreciated!
> 
> Thanks in advance,
> Yassen
> 
> --
> 
> Yassen Damyanov
> Adelie Ltd.
> 
> 
>   
> 
> Looking for last minute shopping deals?  
> Find them fast with Yahoo! Search.  
> http://tools.search.yahoo.com/newsearch/category.php?category=shopping
> 
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-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] BRI card with PCI-E interface

[Olivier]

Hi Alberto,

2008/2/1, Alberto Pastore <[EMAIL PROTECTED]>:
>
> Olivier ha scritto:
> > Hi,
> >
> > Does such card exist ?
> > It seems all existing models are designed for PCI buses.
> >
> > Regards
> >
> >
> > 
> >
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>
> The Sangoma A500BRX is a 2 to 6 bri pci-x interface,
> although I've never tested it
> (A500BRECX comes with hw echo cancellation).


Could you slide it in 1U rack server.
If my memory serves me right, Sangama cards must sometime be stacked
specifically.

Regards,
> Alberto.
>
> --
> Alberto Pastore
> B-Press Srl - Gruppo MSoft
> P.IVA 01697420030
> P.le Lombardia, 4 - 28100 Novara - Italy
> Tel. 0321-499508
> Fax 0321-492974
> http://www.msoft.it
>
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Re: [asterisk-users] "Real" API for Perl?

[Alex Balashov]

Ken D'Ambrosio wrote:

> Hi, all.  I've used the perl/AGI interface, and... well, I found it kind
> of hokey.  Granted, this was in 1.2 days -- perhaps things have changed. 
> Regardless, I guess I have two questions:
> 1) Has the Perl/AGI "binding" improved since then?
> 2) Is there any chance of a "real" API for Perl?

What is your criterion of "real"?  That is to say, what do you need that 
it does not provide?

I've used AGI and FastAGI in Perl extensively and it is yet to fail to 
serve my purposes.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] X-Lite Softphone keeps de-registering?

[Alex Balashov]

Doug wrote:

> The client is travelling much of the time.
> 
> Is there some way that he can use Port 80 so
> that the firewalls that he is behind won't
> block the connection?
> 
> Any other hints or suggestions are very
> welcome!

I suppose the usual answers apply.  You can have Asterisk bind to UDP 
port 80 in addition to port 5060, or you can do destination NAT.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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