On Sat, 20 Jan 2007, Andrew Joakimsen wrote:
Assuming your PRI supports timing from the remote end (CO) which I
highly suspect is the case, then you should set the asterisk machine
to be a slave to the CO timing and then set any other interfaces you
have to NOT be masters, so that the CO
On Sun, 21 Jan 2007, Cristian Draghici wrote:
IDEfisk is based on iaxclient (iaxclient.sourceforge.net) which will
reject a call with the BUSY signal if there is no available line in
the softphone to take the call.
This means you need to configure IDEfisk to use only one line (call
context). I
On Sat, 20 Jan 2007, Samy Antoun wrote:
Hi,
I was wondering if it is possible to connect a skype phone adapter, for
example:
http://zonetusa.com/DispProduct.asp?ProductID=191
http://www.actiontec.com/products/communications/ipw_usb/index.php
Hi list
I encountered problem in using Background command. Below is my
extensions.conf.
[mainmenu]
exten = 4,1,Wait(1)
exten = 4,2,Background(thank-you-for-calling)
exten = 4,3,Goto(n01|s|1)
[n01]
exten = s,1,NoOp(${CONTEXT})
exten = s,2,Background(thank-you-cooperation|m)
exten =
Hello,
Send an email to [EMAIL PROTECTED] i think we the upcoming
version has some fix for this iirc
Zoa
Nir Simionovich wrote:
Hi Philipp,
Thanks for the tip, but that is not what I initially meant. I'm
using IDEfisk, and I would like it when a call comes
Into IDEfisk to generate a
Hi,
Gordon Henderson wrote:
On Sat, 20 Jan 2007, Andrew Joakimsen wrote:
Assuming your PRI supports timing from the remote end (CO) which I
highly suspect is the case, then you should set the asterisk machine
to be a slave to the CO timing and then set any other interfaces you
have to NOT be
Hi everyone!
I just want to thank everybody. My phone works now and just a little hint:
set qualify=no in sip.conf of your phone's extension.
Best regards
Mihaela MJ
On 1/21/07, Token PBX [EMAIL PROTECTED] wrote:
On 1/20/07, Pavel Jezek [EMAIL PROTECTED] wrote:
you have probably
What it actually does is tell the SIP channel driver to track whether or not
any given peer has a call to it. It can then subsequently inform the Queue
application so that another call will not be given to that user. If you did
not have the ringinuse=no in your queue definition, you would
Nothing much to be said.. I backported ringinuse, autofill and the QueueLog
application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't
matter.
They have received minimal testing but appear to function correctly. As always
with these things, don't blame me if they connect your
So it sounds like it should just work. I'll let you know in a few
weeks time :)
TDM400P to E1/T1 card faxing fails by design. The lack of
synchronisation between cards means it can *never* work with any
reliability. The hardware will not permit it.
(And I know I've asked this before, but
I bought a MV-372 for 2 SIM cards as the one channel model seems to work
well (see
http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk).
The setup is such:
- Inet -- VoIP provider --- POTS
|
|
(iax2, NAT)
|
asterisk
(on abox with iptables fw)
|
(SIP,
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