David Backeberg wrote:
On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti
mleone...@evolutionce.com wrote:
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that
Sean Brady wrote:
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
On 02/21/2010 02:02 AM, Kyle Kienapfel wrote:
Hi, I stumbled upon mentions of a SILK codec last night on skypes
skype for sip information page. I tried looking into it further and
found some blog and mailing list posts from 2009 but I can't find any
mentions of anything other than skype using
On Sun, Feb 21, 2010 at 06:56:04AM -0500, Mike A. Leonetti wrote:
Well, when you're right you're right. If it's really that much of a bad
idea I'll just put in for a real machine. Although virtualizing seems
to be all the buzz lately so I was just wondering if I could consolidate
hardware
Hello,
I have asterisk 1.6.0.20 and Is it possible to add Reason header on
Hangup:
Reason: q.850;cause=17
Thanks
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Best Regards,
Giedrius
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Magnus Benngård wrote:
Gentlemen,
I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk.
0851711201 and 0851711290 is on our WAN, no NAT.
0197673581 is outside our WAN and needs to be NAT'ed.
Sending a fax from 0851711201 to 0851711290, no problem, switches to T38
and fax
What I do is have this:
exten = h,1,UserEvent(DialHungUp,ActionID:${CfMC_ActionID} ${UNIQUEID}
${CHANNEL} ${CfMC_AgentToUse} ${CfMC_DialInfo} ${CfMC_QueueToUse}
${HANGUPCAUSE} ${DIALSTATUS})
And then in AMI I know hangupcause and dialstatus.
--
Jim Dickenson
t38pt_usertpsource=yes seems to do the trick, switches to T38 and fax
seems to go through (cant be 100% sure, the fax i am sending to is 500 km
avay from me, but i dont get any errors and my fax thinks everything is ok,
so I cross my fingers),,,
On Sun, 21 Feb 2010 16:36:42 +0100, Johann
21 feb 2010 kl. 16.14 skrev voipas:
Hello,
I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup:
Reason: q.850;cause=17
No, you will have to change the code. I think there's a patch in the bug
tracker. Go search on issues.asterisk.org.
We do add a similar
Hi,
I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system,
up to recently everything
was fine but we are starting to experience the call limitation of the
line (15).
So as to warn user of the problem i attached a vocal notification to the
CONGESTION status after a Dial(),
but it
I'm having trouble with ExternalIVR's socket connection. Is it working in the
current 1.6.0 trunk? I'm getting this error:
Executing [...@ck987_externivr:1] ExternalIVR(SIP/itp_jnctn-006c,
ivr://127.0.0.1:9012) in new stack
[Feb 21 13:38:12] NOTICE[15864]: app_externalivr.c:631 eivr_comm:
Does any one put a HFC-S card working in nt ptp mode?
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Yes, it will catch congestion cases on the receiver side.
There's two ways to avoid it:
1) use ChanIsAvail on the dadhi channel. It works very well.
Or
2) Create a special signal for dahdi congestion (require modifying the
source and recompile) that differs from the congestion cause. Here we
On 19/02/10 8:15 AM, jonas kellens wrote:
How about something like :
[mycontext]
exten = 100,1,NoOp(calling 100)
exten = 100,n,NoOp(going realtime)
switch = Realtime/mycont...@realtime_extensions
mailto:mycont...@realtime_extensions ; from here on we use realtime
And then my MySQL-DB
On 20/02/10 10:53 PM, jonas kellens wrote:
I have read on this list that people do not get a reply if they ask
stupid questions.
Is this then a stupid question that I ask ?
If nobody has ever combined extensions.conf and realtime in a way that I
want to do, I wanna hear it too. Even if this
Hi all,
I've uploaded a free app for the iPhone called AsteriskRef to the Apple
AppStore.
This allows you to lookup applications and functions using your iPhone
or iPod touch so you don't have to jump out of extensions.conf or open
another terminal tab.
It currently supports applications and
Sean Brady wrote:
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
Several people in the past month have asked about sending audio of the call to
a remote AGI server. Currently, this is not available, because when we
connect to a remote AGI server, we connect on a single socket, which
establishes a command channel. I've been thinking about how we might
hi, all
in my test,it shows Playback will answer the call automaticly, but i
don't want to so.
i will use answer function to answer the call. could you help me ?
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Best regards,
Sucan
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On Sun, Feb 21, 2010 at 07:55:39PM +, Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
Which version of Asterisk do you use? Which channel driver?
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Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
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