Displaying 20 results from an estimated 1000 matches similar to: "Dialstatus Oddity in 1.2"
2004 Sep 23
4
Asterisk 1.0 RPMS RH73 and RH9
Hello,
Straight from the floor of Astricon 2004, I am happy to release my
updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform.
Current Release
---------------
asterisk-1.0-0
libpri-1.0-0
zaptel-1.0-0
kernel-module-zaptel-1.0-0
RedHat 7.3
----------
ftp://ftp.nacs.net/asterisk/rh73/RPMS/
ftp://ftp.nacs.net/asterisk/rh73/SRPMS/
RedHat 9.0
----------
2007 Sep 10
2
Failover SIP logic
I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy)
extensions.conf:
[globals]
trunk_1 => SIP/trunk1
2004 Feb 03
0
Asterisk 0.7.1 RPMS Updated to Rel 4
Neo: What are you trying to tell me? That I can dodge bullets?
Morpheus: No, Neo. I'm trying to tell you that when you're ready,
you won't have to.
There have been over 500 downloads of the RedHat Asterisk RPMS
since they were released 2 weeks ago, and I have received many comments
to improve them. After some late night hacking this weekend, I have
dropped 0.7.1 release 4 RPMS at
2004 Jan 24
1
Asterisk RPMS for RH9 + RH7.3
Hello all,
It's my birthday today, so as my present I would like everyone
possible to download and test my updated set of RPMS for Asterisk 0.7.1.
By popular request, I installed and built a set of RPMS for RedHat 9.0,
and in the process fixed a bunch of issues from the initial build. I have
also updated and will be maintaining a page on the Asterisk Wiki located
at:
2005 Jan 22
1
ASTCC: potential billing issue and "fix"
Before I start, I just want to say this is not necessarily a problem
with ASTCC, but may be a problem the way I have set up ASTCC (and
possibly the way others have set it up as well). The issue is that ASTCC
tries to match the pattern *anywhere* in the called number, not
necessarily only at the beginning.
I have set up ASTCC Routes like this:
1800 Tollfree Trunk1 0 0 100
1416 Canada Trunk2 0 0
2004 Jul 17
0
Updated RPMS for Asterisk-1.0 RC1
Hello all,
Hot on the announcement this morning by Mark, I have updated and
rebuilt new Asterisk RPMS for RedHat 7.3, 8, 9 and Fedora Core 1. Please
feel free to install these and beat them up. The usual disclaimer
applies.. These haven't been tested, not reccomended for production use,
don't bug Digium for support of these, blah blah blah....
You can find the updated RPMS and Source
2010 Mar 29
1
Asterisk, IAX, & Sub interfaces
Is there anyway to get the following scenario to work...
I have 3 IAX trunks that I want to setup to peer with other * boxes. I have 1 physical interface, eth0. I also have 2 sub interfaces, eth0:1 & eth0:2. I want to setup a single IAX trunk on each of the interfaces. All 3 interfaces are going to have separate publicly routable IPs, and for this purpose, let's say that because of
2006 Oct 10
0
Cubix / Firefly softphone and Asterisk
Hi All
Has anyone used Cubix / Firefly successfully with Asterisk? When
someone calls a Cubix softphone, Cubix never seems to answer the call
correctly. The other person just hears ringing even though it has been
answered. I am using IAX as the SIP support doesn't seem to 100%
either. Idefisk works 100% on the same setup.
Kind Regards
Garth
2006 Apr 20
2
Cubix Softphone + Asterisk 1.2.6
I've tried Idefisk and Cubix Softphones, and they both work fine, except
for two issues:
1. Idefisk seems to have a longer delay between the time I can hit
tones, and
2. Cubix, while can send DTMF faster, never actually connects to an
Asterisk-dialed call -- I can't hear the party who answers.
#2 has been asked but unanswered here:
2010 Jan 28
3
Dial cellphone from one PBX1 to PBX2? is it possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)
my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
2003 Apr 07
0
Call FWD & the new channel driver chan_local
I just thought i'd post a small sample that uses the new chan_local
to show one way of doing variable callfwding
This sample extension.conf uses's the ast DB to store a users current
extension,
in a db family of CallFWD
and the unique Key is based on the current channel the user is assigned.
In the globals var section each key is hardcoded EXT1, EXT2 this is used in
the
[incoming] context
2004 Apr 17
2
SIP device rings once on busy before giving busy tone with dialplan
Hi!
I am having difficultly in having users of various SIP devices obtain the
correct behaviour when they call a busy number ie. only hearing the
Congestion/Busy tone. I assume this might be because the SIP device
itself generates the 'ring' tone?
With my current setup in the dialplan extract (below) the user of the SIP
device hears one 'ring' and then the busy tone if a number
2004 Apr 29
8
GrandStream 1.0.4.55 Firmware
Hello,
Anyone using the 1.0.4.55 firmware release with any success? I
have had my Budgetone running 1.0.4.50 for about a month and a half now
with no problems whatsoever, and I am a little leary about upgrading.
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
2005 Jul 20
3
[Asterisk-Dev] Memory Leak in Stable?
Hello,
I have a client that has a fairly small installation (20 SIP
Phones) that is running Stable. Asterisk appears to be consuming large
quantities of memory, and growing uncontrollably to the point where after
about 6 weeks the box starts to swap itself to death. I've been keeping my
eye on it today, and in the last 12 hours, it has grown by about 8
megabytes, and there has been
2004 Sep 23
11
1.0 Mirrors
Hello,
Please be conscious of Digium's bandwidth and use a Mirror when
downloading 1.0. I have mirrored the tarballs at:
ftp://ftp.nacs.net/asterisk/
Direct links:
ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz
ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz
ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz
--
Vice President of N2Net, a New Age Consulting Service, Inc.
2005 Jan 29
2
TE405P w/ Intel SE7210TP1_E Motherboard
Hello,
I'm looking at building a couple new PRI Gateway boxes using
TE405P cards, and was wondering if anyone has had any experiences (good or
bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics
builds some really nice (and cost effective) 1U servers based on the
board:
Server: http://www.gtweb.net/gt637.html
Specs:
2010 Jul 12
1
Fax for Asterisk, capable of receiving from website but not from fax machine !!
Hi Guys,
i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1)
and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue
i'm having is that i'm able to receive faxes from a website (that offer this
service) but not able to receive from a regular fax machine (that is working
perfect).
[fax-rx]
exten => receive,1,NoOp(**** FAX RECEIVE ****) exten
2005 Mar 01
1
Connecting Asterisks via SIP
Hi.
It is propbably a really naive problem, but I really couldn't find
answer how to connect two Astrisks via SIP. I managed to do it via IAX
without any problem. But this is a test installation and I would like to
connect them via SIP.
So I have two computers:
pbx1 - 10.1.3.207
pbx2 - 10.1.3.204
pbx1 handles extensions 1xx and pbx2 for extensions 2xx. I would like to
call user from pbx2 to
2005 Jan 12
3
Polycom IP 500 Dial Issues
Hello,
I have a mixture of Polycom SP IP 500 and 300 phones. I have been
reading through the administration manual to try and solve this problem,
but I do not seem to be able to find the answers to my question. I figured
I would ask here and see if anyone has some suggestions.
The problem is kind of annoying. After dialing 4 digits, the phone seems
to pause and miss the 5th digit, often
2006 Nov 06
7
DTMF Tones occuring randomly
Hi,
I have asked this question months ago - i have "toggled down" all DTMF
Recognizations in my Asterisk (no more features etc)
and found more people which recognized the same problem, but i cant find
any help for them and me.
The Problem (short as possible) :
In a randomly call in my business day some unit in my Asterisk System
sends an randomly DTMF Tone, like "A"