Am I correct that if I'm running an -rc or from an SVN release tree
that there's no way I can use any commercial add-ons from Digium, such as
Skype, Cepstral, or G.729?
No, happily not correct. :-)
Digium tries to make their add-ons work with all major releases of
Asterisk. You
You should not try to mix modules for different major versions of
Asterisk. 1.6.0.x modules should only be used with 1.6.0.x, etc.
While John's previous comments were not incorrect, it is unfortunately
quite common that there are API/ABI changes between major releases that
necessitate
I'm using asterisk meetme function like:
exten = 9070,n,MeetMe(|dcM)
and everything works pretty well. But I would like to add a review of
the entered conference number before the user jumps into the conference.
Somthing like:
*:Please enter the conference number followed by the
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
Forgive me for the question, but /why/ do you want this behaviour?
Isn't the whole point of dialling multiple extensions so that a call has
a greater chance of being
I have a SIP phone calling an AGI application. It starts out this way:
-- Executing [...@macro-call-agi:2] AGI(SIP/151-b414f0c8,
computer-temp.sh,darwin,) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh
Then I get a dozen or so copies of:
[Nov 30
What version of Asterisk are you running? This sounds similar to an
issue with AGI's I saw a while ago, but I can't quite remember
exactly what the issue (or issue number) was.
1.6.2.0-rc2
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On a closely related note, has anyone built a normal (not embedded)
system on SSD?
I've been running Asterisk on a 20GB SSD drive for a while now.
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asterisk-users mailing list
What mft/model?
Actually, it's 16GB, not 20GB. It's a Transcend TS16GSSD25S-S.
I know that CF cards have a limited number of writes before frying.
If we keep it from using swap am I really only concerned about
voicemail and logs?
That number is quite large, though. I'm taking backups and
And? Noticed any significant performance advantage?
I never ran it any other way, so have no comparison point. I didn't do it
for performance reasons, but reliability.
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If I have a SIP provider (in this case a PBX using SIP trunks), and
I want to send the calling extension number and name as the from in
the SIP invite, how do I set up my sip.conf entry for that provider? I
find the documentation confusing on this point.
callerid=Some Name In From Header 7065551212
So the first part is the NAME and the second the number, right?
But my question was how to have that be information from the CALLERID
channel variable rather than a fixed value in sip.conf.
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Is this patch correct? The doesn't make logical sense to me. I think
it should be || and making this change fixes the problem I have with SIP
phones in MeetMe conferences. If it's correct, is there someplace more
formal that I should submit it to?
*** app_meetme.c.old2009-10-11
David Backeberg wrote:
From a quick glance at your patch, I would say that it probably tries
to address the audio quality problems I and others were experiencing.
No, it's fixing a much more serious issue. As I sent to this list twice,
when I have a conference between Dahdi ports and SIP
What version are you running?
1.6.2.0-rc2
Does that version support disabling talker optimization?
Yes.
Have you tried disabling talker optimization?
Yes. That's how I found the bug. I got no audio from the SIP phone
into the conference, so I decided I'd try seeing if it did if the SIP
I sent a query about this before, but have some further information and am
hoping somebody has a suggestion as to what to try next to debug this.
I'm using an Asterisk box primarily for MeetMe conferencing. There are
two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works
fine
Robert McGilvray wrote:
You can do this in the dialplan. Just launch MeetMe with different
options based on the caller,
What's confusing me is that when I look in app_meetme.c, the relevant
options are stored in what are called conference flags and there are
separate user flags. This makes it
We've started to use Asterisk for conferencing and have been getting some
complaints. Our configuration is that some people call in from home, but
we have a physical conference room with a Polycom. When somebody was giving
a presentation in the physical conference room, we were told that the
How do I properly quote things when I want to use the IF function on
something returning a string with blanks (e.g, CALLERID(name))?
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AstriCon 2009 - October 13 - 15 Phoenix,
Richard Kenner wrote:
How do I properly quote things when I want to use the IF function on
something returning a string with blanks (e.g, CALLERID(name))?
Use double quotes around your variable
Thanks. That was my second try, but I thought that it didn't work
because I introduced a typo
I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call
from the SV8300, I see:
[Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226 dahdi_pri_error: !! Unknown
IE 50 (cs5, len = 3)
I see an IE 50 in the Q.932 specification, so I don't understand why
this error is occuring.
The two patches attached on the issue apply against libpri branch
1.4 and asterisk trunk, respectively. Both are required. Given
that it's been 5 months since I first created the patches, I have
redone them tonight, in order to facilitate testing.
Thanks!
I checked out that branch with
My system is linked to a legacy PBX via Q-SIG and most of my tests so
far have been from that PBX. I created a number of MeetMe conference rooms
and they work fine when called from the legacy PBX. However, when there's
a MeetMe room with a legacy caller and a SIP phone, the SIP phone can
hear
I'm using QSIG between an NEC SV8300 and Asterisk (after giving up with
CCIS). Things work pretty well with the exception of issues on stations
on the SV8300.
When I call from Asterisk to a SV8300 station and I send my extension
as the caller ID number, it shows up on the SV8300 as OPERATOR.
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