Displaying 20 results from an estimated 2000 matches similar to: "incoming limit, call_limit, or call-limit?"
2006 Apr 08
6
How to set busy
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up from there.
What would be really great is if I could control how many calls by the
context. So if a call was routed via
[overload] Then the ext wouldn't report busy it would just keep ringing
available
2006 Mar 04
1
*** Yet another boring weekend? Test new Asterisk features in development!
In Sweden, where I live, it's snowing like crazy. The Stockholm area
is covered in white stuff
and there's really no reason to leave the computer and get out
anywhere. More white stuff
is coming down all the time. Boring. I am sure your weekend is no
better - rain, snow or
just another boring sunny day.
Let's find something cool to do during this weekend!
Join the cool crowd
2003 Oct 20
3
Call Waiting on SIP phones
Hi All,
This is the first time I'm submitting a patch, and I hope it fixes more than
it breaks. I'm putting it here, since John Todd mentioned a while ago about
the heavy load Mark and crew have at Digium (doing such good work), so I
thought all of us could test this first, and if ok submit for inclusion in
CVS later if appropriate.
This is an extension to work done earlier (sorry I
2006 Jun 20
2
Call limit function on sip channel to external pop
Hi,
We've been using asterisk as our main telephone-communications platform
for years now, and we wrote several extra scripts and features for it.
Now we 're looking for a solution to limit the number of channels going
to an external SIP provider.
We recently upgraded our system from asterisk 1.0 to 1.2(.9 now) to be
able to use such features, but nothing helped...
When we configure a new
2006 Feb 21
1
Test my test-branch!
Friends,
The developer team for Asterisk not only consists of coders - a very
important part are the testers, those that test new code and give
feedback.
For a few weeks, I've been maintaining a large number of branches
with various stuff in them and have gotten very little feedback, not
enough to judge whether or not to move forward with these patches.
Some, but not all, code is
2003 Aug 07
1
Sip Trunk config
incominglimit is already implemented for SIP. Just specify under the
endpoint how many incoming connections are allowed.
For example,
[cisco]
type=friend
username=cisco
secret=blah
nat=yes ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms away
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 =>
2011 Mar 29
1
wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my
ten digit "DID". I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is "asterisk".
Is this a bug? Or is there some other config I must make ?
register = 2155551941:123456 at 10.0.138.226/2155551941~600
[peer](!)
type=peer
context=inbound
qualify=yes
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 Apr 12
1
Asterisk (1.4) and hints/presence/BLF
Playing with hints/presence/BLF on asterisk I've made the following
"discoveries".
1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says:
"If you add incominglimit=1 to your peer in sip.conf, the SIP
channel will notify you when that extension is busy."
As "incominglimit" is obsolete you can use "call-limit".
2004 May 22
1
app_queue and app_groupcount
The new app_groupcount looks great for most applications but it a is a
step back for call queueing...
since app_queue calls physical interfaces and not extensions,
app_groupcont can't be used to limit the calls passed to a dynamically
added agent.
I presently use the broken sip incominglimit feature (even though it's
less than ideal as it also limits outgoing calls preventing
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
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
2004 Apr 20
3
Limiting incoming SIP calls & Original CallerID on transfer
I have 2 issues which I need to resolve on our production Asterisk
server:
We are currently using Polycom IP600 VOIP phones for our office which
are capable of handling 2 calls per SIP registration. What we're finding
is when staff are on the phone, Asterisk will pass them a second call
which will show up on their display, and an audible beep is heard over
the phone (regular call waiting). I
2003 Dec 02
2
incominglimit stuck in app_queue
Hello,
Right now I have app queue working with incominglimit=1, there is no
call waiting signal, but after a while( like couple of hours) some
phones randomly get stuck. The * thinks that they are in use and doesnt
ring them, when they are infact not in use.
sip show inuse, shows that they are inuse. typing reload on the console
resets this and they are again available for working.
anybody
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 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
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
2003 Nov 17
2
Hunt groups and SIP?
I would like to setup a hunt group, not a group ring, using sip phones.
Anyone done this with sip devices? Comments suggestions?
I have not had much luck with the outgoinglimit=1, incominglimit=1
stuff that I would need to get busy extinctions to work right, which is
why I'm asking on the list.