Displaying 20 results from an estimated 20000 matches similar to: "Re: [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT ***"
2005 Jul 15
1
[Asterisk-Dev] call pickup with snom function keys now working with cvs-head + patch sipsubscribe-20050715.rev779.txt
hi listmembers,
please test my new patch to chan_sip.c which is to make call pickup on
the snom phones (and maybe other phones that support 'INVITE/Replaces')
work and make comments in the bugtracker
http://bugs.digium.com/view.php?id=3644 so it can make its way into the cvs.
this patch sipsubscribe-20050715.rev779.txt enables:
* monitoring of other lines (using the 'hint'
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 23
0
Re: Subscription state after reload (New subject)
*lol* I think he was referring to a ASTERISK reboot, not a phone reboot.
> -----Original Message-----
> From: Olle E Johansson [mailto:oej@edvina.net]
> Sent: Thursday, March 23, 2006 1:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Re: Subscription state after reload (New
> subject)
>
>
>
> 23 mar 2006 kl.
2006 Mar 23
0
Re: Subscription state after reload (New subject)
How can a reload clear registrations?
If I 'reload' without using realtime, I keep my sip peers (as well as astdb). I can still contact other phones... registration info is still there.
If I `reload` with realtime, I lose my sip peers (but astdb remains). I can _STILL_ contact other phones.... registration info is still there and Asterisk must be referring to astdb to find the IP
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
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
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
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 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
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
2007 May 29
4
*End Of Life ASTERISK 1.2.X Was: INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
Rather hasty I think. I think whatever version 1.2.X winds up on should
be the most stable release of Asterisk, period.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Olle E Johansson
> Sent: Tuesday, May 29, 2007 5:19 AM
>
2003 Oct 27
0
Fwd: Re: Asterisk on FreeBSD
--- "Olle E. Johansson" <oej@edvina.net> wrote:
> From: "Olle E. Johansson" <oej@edvina.net>
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Asterisk on FreeBSD
> Date: Mon, 27 Oct 2003 08:24:22 +0100
>
> Rich Adamson wrote:
>
> >>My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD
> server.
2006 Jun 08
0
SV: Using regcontext
Hello
Thanks for the answer... Just realized it myself, as your mail arrived :)
Could be a nice feature though.
Jon
-----Oprindelig meddelelse-----
Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Olle E Johansson
Sendt: 8. juni 2006 12:09
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] Using
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 May 01
0
Re: [asterisk-dev] SRTP implementation
> Olle E Johansson wrote:
>>
>> 23 apr 2007 kl. 19.55 skrev Russell Bryant:
>>
>>> John Todd wrote:
>>>> To morph this into a -dev thread: if this patch were to become (again)
>>>> useful and error-free, is there any objection or usefulness in adding it
>>>> to TRUNK? Personally, I think there is, if there is a method by which
2004 Jan 08
2
SIP reload configuration problem /* New subject */
When creating users in the sip.conf file, they do not appear when running the "sip show users" command from the CLI until i restart. A reload doesnt make them appear.
As i said, I am new to the whole Asterisk thing, however have worked with IP/SIP PBX's for a few years - its most likely a user problem though!
Check it out and let me know what you get.
Cheers
Chris
PS - I would try
2004 Apr 15
3
* Announcement * Astricon 2004 - call for speakers!
We're proud to announce Astricon 2004 - the first Asterisk user's
and developer's conference!
* Where? Atlanta, USA
* When? September 22-24, 2004
The conference is arranged in partnership with Digium.inc and the keynote speaker is
Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers
already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara
2010 Nov 24
0
IPv6: What You Need to Know Now
Yes, the thanksgiving holiday is here (in the USA)! But also, the fear
of running out of IP addresses next year has raised its ugly head and
since we don't do Thanksgiving in Europe, we have some serious talking
to do about this problem.
This Friday at 12 Noon EST, Olle Johansson will be joining us to
describe the state of the migration to IP v6 in VoIP-dom. Olle (@oej)
needs no introduction.
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 May 19
1
Development news :: Smarter medialess calls!
Friends,
To update you on recent changes in svn trunk, I can inform you that
we now have ever smarter
ways to handle media streams in Asterisk than we do in 1.2 for the
IAX2 and SIP protocols.
* IAX2: Splitting signalling and media apart
Starting with the IAX2 protocol, we now have the ability to transfer
media streams to go directly
between IAX2 servers and keep the signalling path.