Michael wrote:
> I want to obtained the called number (aka DID) as a variable. How is this
> done?
>
> (Upline connectivity is via SIP provider).
>
> With verbose = 100, the called number is clearly shown as
> follows "Wait("SIP/[called number]-64e6", "1")" for example, but I can't find
> the in
I want to obtained the called number (aka DID) as a variable. How is this
done?
(Upline connectivity is via SIP provider).
With verbose = 100, the called number is clearly shown as
follows "Wait("SIP/[called number]-64e6", "1")" for example, but I can't find
the info as to how to get it as a v
Try two.
On Sun, Feb 22, 2009 at 9:11 PM, Daniel - Asterisk wrote:
> Hi,
>
> I've just installed DAHDI at two PBXs as follows:
>
> *PBX-1PBX-2*
> FXO - FXS
>
> When I try to send calls from PBX-1 to PBX-2 I just receive the message:
> "Starting simple switch on 'DAHDI/
Michael Graves wrote:
> On Sat, 28 Feb 2009 14:59:23 -0800, Paul Chambers wrote:
>
>> Michael Graves wrote:
>>
>>> Witness the fact that the old Pingtel phones ran Java, and they were
>>> incredibly lame.
>>>
>>> I think part of what this thread misses is that DSP is a god chunk of
>>> wh
On Sat, 28 Feb 2009 14:59:23 -0800, Paul Chambers wrote:
>Michael Graves wrote:
>> Witness the fact that the old Pingtel phones ran Java, and they were
>> incredibly lame.
>>
>> I think part of what this thread misses is that DSP is a god chunk of
>> what SIP phones need. A general purpose CPU i
Michael Graves wrote:
> Witness the fact that the old Pingtel phones ran Java, and they were
> incredibly lame.
>
> I think part of what this thread misses is that DSP is a god chunk of
> what SIP phones need. A general purpose CPU is not the right tool for
> the task. A cheap DSP is better suit
On Sat, Feb 28, 2009 at 11:24:53AM -0800, Michael Higgins wrote:
>
> So, tweaking configs, rebuilding this and that... restarting, twiddling, it
> works (yeah!), but fails on re-boot to work at all. Consistently, though.
>
> I believe it comes down to this: I can call out only *after* I've recei
So, tweaking configs, rebuilding this and that... restarting, twiddling, it
works (yeah!), but fails on re-boot to work at all. Consistently, though.
I believe it comes down to this: I can call out only *after* I've received a
call.
So, cold boot. Then:
modprobe dahdi
modprobe wctc4xxp
On Fri, 27 Feb 2009 16:41:16 -0500
Doug Lytle wrote:
> Michael Higgins wrote:
> > exten => _X.,1,Dial(DAHDI/1,${EXTEN})
> >
> >
>
> This should be _X.,1,Dial(DAHDI/g1/${EXTEN})
>
> Doug
>
>
Thanks for the help. Indeed it needs to be:
exten => _X.,1,Dial(DAHDI/1/${EXTEN})
I don't know wh
Sono temporaneamente assente. Rispondero' ai Vostri messaggi appena
possibile.Cordiali saluti.
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Hi all
I'm using asterisk for making PSTN calls from extensions registered on
OpenSIPS...In peak hours ,number of calls Increase dramatically to a non
logic number..When checking the calls using asterisk CLI I saw a lot of
calls in ringing status and after 300s(rtphold timeout), asterisk releas
If this IS back-to-back NAT, it's a science project (AKA too hard). All
this DMZ and port forwarding/DNAT and ALG crap isn't worth it.
Just use OpenVPN.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1
Gary wrote:
>
> On the router, I've turned on "DMZ" to point to my Asterisk box's static IP
> address.
>
> My home (real world) IP address is static.
>
> The Problem: When I grab one of my Cisco 7940's and take it to my office,
> it does not "see" or "register" with my home Asterisk server afte
On Sat, Feb 28, 2009 at 9:53 AM, Gary wrote:
> I've been reading this forum for over the past 4 years and have gained a
> wealth of knowledge. - Thanks to all!
>
> I don't post very often but I've just ran into a problem/condition that I
> simply can't figure out. - Hopefully some kind soul will
I've been reading this forum for over the past 4 years and have gained a
wealth of knowledge. - Thanks to all!
I don't post very often but I've just ran into a problem/condition that I
simply can't figure out. - Hopefully some kind soul will help me.
I've got an Asterisk server in a lab environme
On Fri, Feb 27, 2009 at 2:45 PM, Daniel Hazelbaker
wrote:
> Is there a way to force a channel to continue in the dialplan after
> the remote end hangs up?
You use the 'h' side of the dialplan for the extension.
exten => s,1,Answer
exten => s,n,Set(some magic to make the filename unique)
I have just acquired an eicon diva server BRI-2M card for using with asterisk
and hylafax.
Of course i installed it in a new server with Debian Lenny wich has a kernel
2.6.26+
I have downloaded the latest drivers (9.0) from the Eicon web site and i'm
having trouble
compiling them, specifically
Ne manque t il pas des espaces entre } > 24]
- Original Message -
From: Jean-Michel Hiver
To: Asterisk Users Mailing List - Non-Commercial Discussion ;
klaus.mailingli...@pernau.at
Sent: Friday, February 27, 2009 2:01 PM
Subject: [asterisk-users] [FIXED] Re:
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