Displaying 20 results from an estimated 10000 matches similar to: "Anonymous sip register"
2008 Feb 13
3
What is a "secure call"?
Friends,
The following mail was sent earlier to asterisk-dev and did not cause
the amount of discussion I hoped it would.
Now that we have a way to secure signalling in IAX2 and SIP in
Asterisk svn trunk, we need to start working on
the concept of a "secure call" - or does it really matter?
In SIP, there's a specification for how I as a domain owner can
request all calls to
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 =>
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
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 Mar 07
3
indications & SIP
Apologies if this is an old question; I've searched the list and the
wiki but have not been able to find a definitive answer.
I have an Aastra 480i phone registered with * 1.2.4; I want to generate
UK ringback tones when the handset dials another internal extension. On
my Zap channels, I have this in place by editing /etc/zaptel.conf;
however I've had no luck with the Sip handset (I have
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
Abra sua conta no Yahoo!
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,
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
2004 Jul 05
2
Again Sip Registration Fail
Recently I wrote about this problem,
but it still exist and I can't dial my Xlite SIP Phone
So here is the Notice
Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request:
Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for
'10.1.1.11'
The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in
the same network
Here is part from sip
2007 Feb 21
2
How does Asterisk use SIP info command
What Asterisk command I can use to send a SIP INFO command? Thanks for
pointers.
Yuan Liu
2006 Jun 25
5
Signaling and media
Hi List,
Is there a way to tell asterisk to only accept SIP streams from the same
IP address that is used for signaling?
Thanks,
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
D?couvrez la R?union des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
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 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
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 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 Sep 03
3
GTalk functionality Asterisk
Hello
Previous context :- After Looking up sip and IAX2 that require
configuration at router level which may cause some problems like connection
break etc... so i left them ......... and start wondering if there is some
thing that dont require configuration at router layer. The task to
accomplish to make and recieve calls from outside local network using any
protocol whose soft phones are
2006 Mar 07
7
res_mysql.conf & DNS SRV lookup
Hi friends,
I am using Real Time Asterisk Architecture where I have put the
Sip users/peers and extensions defining the dialplan in tables in
a mysql database.
Currently, asterisk points to my single database server as configured:
------------------------------------------
/etc/asterisk/res_mysql.conf
------------------------------------------
[general]
dbhost = xxx
dbname =
2004 Sep 13
2
Sip Outbound Proxy
How do you configure an outbound proxy for Asterisk? If the sip call is
not local I want everything to go to a designated sip proxy.
Thanks,
Chad
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2004 Apr 19
3
One, två, tre, quatre, cinq ... International numbers in say.c
http://bugs.digium.com/bug_view_page.php?bug_id=0001429
* Support for other language syntaxes in saynumber
Accidentally I opened this can of worms to see if we can add support
for other language syntaxes for saying numbers. Seems like Swedish,
english and norwegian follow the same syntax. I've integrated
existing patches for french, danish and soon portuguese syntax.
The steps we're