Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem
http://sourceforge.net/tracker/?func=detailaid=3337581group_id=152230atid=783657
Has anyone been successful when integrating latest version of Asterisk (10
or 1.8,
Op 08-10-12 09:24, Olivier schreef:
Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem
http://sourceforge.net/tracker/?func=detailaid=3337581group_id=152230atid=783657
Has anyone been successful when
I will make an example:
A is an fxs phone with callwaiting=yes in chan_dahdi.conf
X calls A. A answers.
Y calls A. A hears the call waiting tone.
Now if A hangs up before X, then A rings again (which is what I want).
BUT if X hangs up first, then A automatically answers Y without even
ringing.
2012/10/8 Michel Verbraak mic...@verbraak.org
Op 08-10-12 09:24, Olivier schreef:
Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem
2012/10/8 Michel Verbraak mic...@verbraak.org
Op 08-10-12 09:24, Olivier schreef:
Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem
How many fax and voice calls (which codecs for tha latter ones ?) are on
average using your DSL line ?
1. Previously, I experienced failures during the process of converting
incoming PDF documents into ready-to-send fax image files while the reverse
process (from a fax file into a PDF or whatever
Hello,
I have a local Asterisk server that keep loosing its registration to main
Asterisk server. The local asterisk server is on the local subnet, it acts
as a client with extension 808.
Local server
Sip.conf
register = 808:passw...@as2.x.com
registertimeout=20
registerattempts=10
Main
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Monday, October 08, 2012 12:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Sip registration
The Asterisk Development Team has announced the release of Asterisk 1.8.17.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.17.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 10.9.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.9.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 11.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
All interested users of Asterisk are encouraged to participate in the
Asterisk 11
Asterisk 1.8
(a) We will have a group of 4 analog lines into a Digium card that will
be used for local calls. What is the best way to use those lines as a
pool for outbound calls? Can I use ChanIsAvail(), listing those 4
channels, and then use the first one returned?
(b) For emergency
One suggestion I have:
Would it be helpful to know the revision number of rc1 in the release
notes?
I'm on a patched version of Asterisk from Doubango to deal with Chrome's
non-standard ICE candidates, and unless this is included in rc1 (meaning
rc1 is newer than what I have and also deals with
James Mortensen wrote:
One suggestion I have:
Would it be helpful to know the revision number of rc1 in the release
notes?
I'm on a patched version of Asterisk from Doubango to deal with Chrome's
non-standard ICE candidates, and unless this is included in rc1 (meaning
rc1 is newer than what I
After upgrading to Asterisk 1.8.15.1
I'm constantly getting this error on the command line:
ERROR[2499]: iax2-provision.c:266 iax_provision_version: ast_db_get failed to
retrieve iax/provisioning/cache
Can somebody explain what it is and how to fix it?
--
Joseph
--
Can someone refresh my memory how blocking incoming call works based on caller
ID in Asterisk 1.8?
If I remember correctly in asterisk 1.4 it was possible to block caller ID from
the command line, asterisk had some internal database I think.
--
Joseph
--
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