Displaying 20 results from an estimated 10000 matches similar to: "CPC on Zaptel"
2008 Feb 02
3
Zaptel timer on Intel Dual Core servers
Friends,
I'm having severe problems with zaptel timers on Intel Dual Core
systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM
or PRI cards - all ends up with large timer probems - zttest going
down to 50% accuracy on some systems, even to -1 on ztdummy systems
and voice quality is no more. A restart is the only way to get back
to a working system.
We're only
2006 Jan 25
14
No audio? Update your Asterisk
This morning we discovered a serious bug that stopped all bridged audio
in our Asterisk servers. Mark found the problem and soon fixed it.
If you get this problem today, please update your Asterisk server. A fix
has been commited to the subversion repository for 1.2 as well as trunk.
A fixed 1.2.3 release will be published on ftp.digium.com as soon as we
can find a release engineer (consider
2004 Nov 30
0
E1s ISDN PRI & CPC
Hi everyone,
I'd like to try to use * & ASTCC to create a pre-paid public call card
solution. In testing is looks good. I've been talking with Telecom
suppliers about supplying me with E1 Primary ISDN lines (probably 4 to start
with) and I'd purchase digium TE405Ps to connect everything up.
What I'm uncertain about is how to handle CPC with asterisk. I know that
2mbit E1
2006 Feb 19
0
Viking CPC-Disconnect
Someone on the list a while back suggested that if you were having
problems with call disconnects, to look into a product from Viking
TellecomSolutions called cpc-disconnect:
http://www.vikingtelecomsolutions.com/catalog/model_CPC-1.htm
I received my unit on Friday and put it into place Saturday afternoon
(SBC in this area doesn't supply call disconnect supervision). The unit
was acting
2003 Oct 27
1
Fwd: Re: Asterisk on FreeBSD
Your log file almost looks like a bug in Asterisk doesn't it?
Why call poll() with a zero timeout while passing only one FD?
and then why do the read when there is no data?
Read the man pages for all the system calls
Take a look at the source chan_sip.c
/* Wait for sched or io */
res = ast_sched_wait(sched);
if ((res < 0) || (res > 1000))
2006 Mar 24
1
Re: Subscription state after reload (New subject)
Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own.
> Thanks Olle,
>
> So am I to understand that you
2006 Apr 05
5
Dial Plan Logic Problem
Hi
I can't for the life of me work out why this is not
working. When in the campon contect if you hit a DTMF
key 2 you get moved to the exten => 2 defined in the
mainmenu context not the exten => 2 defined in the
campon context. What is wrong? The same happens if you
hit key 1.
[campon]
exten => _*1XXX,1,Answer
exten => _*1XXX,2,SetCallerID(${CALLERIDNUM})
exten =>
2004 Dec 19
3
[Asterisk-Dev] RE: [Asterisk-biz] Asterisk training andcertification :: AstriconTraining
I feel this is a slap in the face for those of us that have been here and I
don't feel I should HAVE to pay to be certified... I think me and MANY
others are about to walk out of the project over this. I have already
spoken with many people that are close to the project. You're hurting US
and our ability to make money. I still know the code better than most of
the people that will be
2004 Apr 20
1
Re: SIP re-invite
Trouble getting chan_sip2 to compile
below is what I have done
-download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software
cp /root/software/chan_sip2s.c /usr/src/asterisk/channels
cd /usr/src/asterisk/
patch -p0 acl.c /root/software/acl.c.patch
cd /usr/src/asterisk/include/asterisk
patch -p0 acl.h /root/software/acl.h.patch
- added the follow to /usr/src/asterisk/channels/Makefile
2007 Dec 15
17
Upgrade to Asterisk 1.4 - it's one year's old!
Friends in the Asterisk community,
I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2
and 1.4 there's been a lot of
important development. New code cleanups, optimization, new functions.
I realize that 1.4 at release time wasn't ready for release, but we've
spent one year polishing it,
working hard with bug fixes. The 1.4 that is in distribution now is
2009 Apr 01
10
FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND
VIDEO TO MICROBLOGGING!
In a surprising move, Digium in partnership with Edvina today released
a new channel driver for Asterisk, chan_tweet. The driver connects
seamlessly to several microblogging platforms, including Twitter,
Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of
this new module is to
2006 Oct 11
4
Psst... Top secret information: Codename Pineapple
Friends in the Asterisk community,
I've been talking for years about the new version of the SIP channel.
I've been trying to get funding
and get going. Well, the funding part remains to be handled, but I
have other news - if you kan keep
it to yourself.
...I've began coding. Finally.
With a happy smile on my face I removed "pedantic=yes" the other day.
After years of
2004 Dec 19
0
RE: [Asterisk-biz] Asterisk training and certification :: AstriconTraining
I feel this is a slap in the face for those of us that have been here and I
don't feel I should HAVE to pay to be certified... I think me and MANY
others are about to walk out of the project over this. I have already
spoken with many people that are close to the project. You're hurting US
and our ability to make money. I still know the code better than most of
the people that will be
2009 Feb 16
1
The cpc provider cometh
Hi,
Finally, the long promised CPU performance counter provider has
arrived into Nevada build 109! For those that need an injection of
excitement at the start of the week, you can check out some usage
examples on my blog:
http://blogs.sun.com/jonh/entry/finally_dtrace_meets_the_cpu
Also, you can find a new section for the provider in the DTrace docs:
2004 Dec 09
1
can FXS ports on TDM400P provide Battery Reversal or CPC
Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy
PBX can be outfitted with caller-ID and is already able to handle Calling
Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid
PBX.
My question is how would one enable Asterisk to control the TDM400P/FXS port
to provide to the /FXO CO port on the legacy PBX, support for proper answer
supervision/CPC
2006 Mar 10
3
Development news :: T38 passthrough support
Friends in the Asterisk.org community,
There is a lot of cool stuff going on in Asterisk development, things
that will change Asterisk and
make it work better in your organisation, make it easier to sell in
your area or give you more
consulting oppurtunities - in short, functionality that will make a
lot of sense for you users.
However, developers can't really get anywhere without a
2009 Aug 25
6
Breaking news, but what happened? 11.000 channels on one server
Hello Asterisk users around the world!
Recently, I have been working with pretty large Asterisk
installations. 300 servers running Asterisk and Kamailio (OpenSER).
Replacing large Nortel systems with just a few tiny boxes and other
interesting solutions. Testing has been a large part of these
projects. How much can we put into one Asterisk box? Calls per euro
invested matters.
So far,
2006 May 31
0
Asterisk Bootcamp in Europe :: June 12-16 and the Asterisk SIP Masterclass in Chicago, July 2006
** Asterisk Bootcamp in Stockholm, Sweden
The next Asterisk Training is the Edvina.net Asterisk Bootcamp - the
class we have been giving for over a year under the brand name
"Astricon Training". The same teacher, the same material and a new name.
All students have a PC and will install a fully working Asterisk PBX.
During the week, we will build a business PBX configuration as
2006 Jun 06
0
What to do on a national celebration day? Test, test, test!
Today is Swedens national day - since a few years a holiday too.
We don't have a tradition on how to celebrate.
Sweden has not been to war for a very long time, so there's no real
spirit
for the country here - it's been aroundfor such a long time, so
what? :-)
Guess we have to learn from abroad, to get a celebration feeling like
July 4th in the US or May 17th
in Norway (from
2008 Apr 13
1
Similar option as promiscredir to use in transfer (REFER)
I made a similar question in a previous thread, but there was no
answer, so I think I was not very clear making the question. What I
need is some configuration that works like "promiscredir=yes" in
sip.conf that enables me to do the same thing with transfer (REFER),
letting me transfer a sip call to a non local sip address.
Thanks in advance,
Thiago
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