Displaying 20 results from an estimated 3000 matches similar to: "Allow INVITE for hold to pass through"
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 27
1
SIP Channel jitter buffer issue
Hi,
I have a SIP client which is registered to asterisk. Asterisk is
registered to a SIP trunk and also handles the media. Now since my client
has some issues in its RTP Tx, which seems to have some amount of jitter
(mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and
max delta is 85 ms), to over come that I have enabled jitter buffer in the
SIP channel by setting sip.conf
2008 Jan 31
1
Incoming call from SIP proxy to asterisk
Hi,
I have asterisk register two users (client-1, client-2) with a SIP proxy.
I have the same two SIP client registered with asterisk. Now my dial plan
setup is such that any call from client-1/client-2 is forwarded to the SIP
proxy and the SIP proxy then takes the routing decision. Calls coming from
SIP proxy will dial out the respective user. Asterisk is required to stay in
the signaling as
2006 Jun 09
1
Sip transfer, Sip on hold
Hi everybody, sorry for my english but i'm italian and i don't know it
very well.
I'm trying to do a java-program to traduce and notify asterisk events to
a Tapi program.
I've a problem with call trasfer.
When i transfer a sip user i would like to put his line on hold but i
can't do it. He listen the music on hold but his state is connected and
not Hold.
I hope you understand
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 =>
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
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
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 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
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!
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
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
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))
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 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
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 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
2014 Nov 24
3
[LLVMdev] bx instruction getting generated in arm assembly for O1
Hi Mayur,
> On 24 Nov 2014, at 07:00, MAYUR PANDEY <mayur.p at samsung.com> wrote:
> In the assembly generated with O0, we are getting the "blx" instruction whereas with O1 we get "bx" (in 3.4.2 we used to get "blx" for both O0 and O1).
>
> Is this because of this patch: [llvm] r214959 - ARM: do not generate BLX instructions on Cortex-M CPUs
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
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