Try moving those into [default]
On Thu, 2005-04-07 at 16:35 +0200, Paul Hewlett wrote:
> exten => i,1,Playback(invalid)
> exten => i,2,Goto(s|2)
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Great info Matt.
Thanks.
/Danny
On Thu, 2005-04-07 at 10:20 -0400, mattf wrote:
> My Sangoma Experience in Asterisk:2005-04-07
>
> MATT---
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Yupp can confirm that, with latest SPA firmware...
Any ideas?
/Danny
On Thu, 2005-04-07 at 02:03 -0500, Brian Capouch wrote:
> Major good news intermingled with the bad: a fresh CVS install tonight
> does NOT crash my MIPSEL-based WRT54GS instance of Asterisk.
>
> However, something fishy is goi
Hi all,
Anyone set up * <-> ipcb.net ?
Feel free to contact me off list for some conf examples, cant seam to
get it right ;)
Regards
/Danny
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Steve Kann wrote:
Paul Fielding wrote:
So far in my playing with Asterisk I've messed with soft phones
(x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters
(Grandstream 286, Digium IAXy).
I've also got a Vonage line, using a Linksys ATA.
None of the devices I've connected to my A
Works like a charm, with a few quirks ;)
/Danny
On Monday 07 February 2005 07.29, Richard Hamnett wrote:
> Hi all,
>
> I've just released an ALPHA version of an application I have been
> working on to help integrate the desktop with asterisk. A list of the
> key features are as follows:
>
> *
B. ffs!
/Danny
On Tue, 2005-05-17 at 22:39 -0500, Brian Capouch wrote:
> Chris Mason wrote:
> > "I have gotten"
> > What language is that?
> >
>
> Found in an English dictionary:
>
> get
> v. got, (gt) gotÂten, (gtn)
> v. tr.
>
> You don't like the rules?
>
> B.
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I'd be happy to host you in our Montreal Datacenter at no cost.
Contact me off-list if you're interested.
/Danny
Matt Riddell wrote:
Andres wrote:
So it looks like Livevoip went Bankrupt
Sh1t.
Looks like the Daily Asterisk News will need a new host.
So, unless anyone can donate space for
also registered spandsp.org,
> so that name works too.
>
> Regards,
> Steve
>
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Thanks for the look see KP ;)
--
Many Regards.
Danny Froberg
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Just a note, if you go with IP tech oversize the UPS bit, since it might be a
good idea to be able to use the PA / Phones incase of power outtage ;)
--
Many Regards.
Danny Froberg
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Dont forget to complement the wiki info if you see anything missing.
Many Regards.
Danny Froberg
On Saturday 19 March 2005 13.53, John Breeden wrote:
> My Cisco 7960G is no longer a killer doorstop! It's now a fully
> functioning phone! Very cool. Thanks to all.
>
> Turns
Word of your booth came back faster than your mail ;)
Only good things where said ;)
/Danny
On Tuesday 22 March 2005 04.06, Thilo Rößler wrote:
> For all who are interested: A quick review of CeBIT 2005. :-)
> CeBIT was a very successfull event. Most of the time, the asterisk-booth
> was crowded w
my day ;)
--
Many Regards.
Danny Froberg
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Try playing with these;
exten => s,n,DigitTimeout(3); Set Digit Timeout to 3 seconds
exten => s,n,ResponseTimeout(20); Set Response Timeout to 20 seconds
Many Regards.
Danny Froberg
On Thursday 24 March 2005 09.24, Joel Jn-Francois wrote:
> > > 1) When an inco
Anyone tried this on Virtuozzo?
/Danny
On Mon, 2005-04-04 at 22:04 -0700, snacktime wrote:
> I just got a linode account and got * up and running without any
> problems. I was going to ask them to load zaptel/ztdummy, but I was
> wondering if anyone else was interested in an * friendly UML hosti
sip reload
or just reload
On Wed, 2005-04-06 at 16:31 +0800, Abraham WEI wrote:
> I modified /etc/asterisk/sip.conf to add my X-Lite soft UA, which is
> assigned the user name "177210". Now I want to add another UA with a
> user name "177209". Well, asterisk is running. How can I make the
> change
19 Question + this one and no answer;
Does anyone have a clue what causes "Unknown RTP codec 72 received" notice
and how to fix it?
Regards
Danny
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T
Think i solved that one by the ordering of allow= in sip.conf
At 18:39 2004-10-17, you wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=0002519
If anyone has seen that error please come forward and report on this bug
please. The original reporter is unwilling or unmotivated to even make an
e
snippet somewhere in their toolbox :)
Would be great so we don't have to reinvent the wheel ;)
Since the communication would be at (V.21) 110,300,1200 bps i think a
software only module would do it.
Anyone willing to give a hand?
Many regards
Danny Fr
e the communication would be at (V.21) 110,300,1200 bps i think a
software only module would do it.
Anyone willing to give a hand?
Many regards
Danny Froberg
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Hi Darren,
It is today, check the variables CALLERID, CALLERIDNUM & CALLERIDNAME
/Danny
At 15:58 2004-10-18, you wrote:
Is there a way to get the Call ID off of a call that runs through *
without loading any kind of billing CDR platform?
If not, I think it would be a great addition to * if the Ca
Rusty,
You do defenitely not want the X100P it's discontinued and rightly so,
horrible card.
/Danny
Rusty Dekema wrote:
Hi,
I apologize in advance if this is a stupid question, but I have not been
able to find an answer by searching the web.
I would like to add an FXO port or two to my A
Howdy,
have working realtime queues using queue_members looking something like;
queuea|Local/[EMAIL PROTECTED]|0
queuea|Local/[EMAIL PROTECTED]|1
queuea|Local/[EMAIL PROTECTED]|10
Regardless of what strategy is used in the queues
(roundrobin,rrmemory,ringall etc) it wont escalate on NOANSWER
Thanks for clearing that up Kevin.
Now on to figure out how to "PauseQueueMember" when enough NOANSWER's
has been detected so he don't fubar the entire queue.
Would be alot cleaner than sending callers to ever higher level queues
*sigh*
Kevin P. Fleming wrote:
Regardless of what strategy is us
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