I just want to make two specific sip phone sets to ring together, when
someone dials a specific incoming extension. And then, when each of the
ringed sets answers, to be placed immediately into meeting session with
the caller together with the other phone set.
Here is exactly what I mean:
Person
Not info about networl settings. Please give output of
ip l
ip -4 a
ip ro
2011/11/1 Danny Nicholas da...@debsinc.com
Hello listers,
Another opportunity presents itself in my 1.4 to
10.0 conversion. My asterisk is set up for 192.168.23.xx and most of my
phones
Sorry it took me a while, but I was ill for a few days J
Part1: http://pastebin.com/SZqgxh7B
Part2: http://pastebin.com/gfJtVVRE
In this log a call from extension 346 is made to queue 900.
Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.
Queue 901 has 1 agent
Good morning,
I have not solved this problem yet, but, I found that the source of
the problem are my macros. For example, I have this context:
context ramais {
101 = dial_sip(exten1);
102 = dial_sip(exten2);
103 = dial_sip(exten3);
};
All these extensions use the
IP outputs
ip 1
Object 1 is unknown, try ip help.
ip -4 a
1: lo: LOOPBACK,UP,LOWER_UP mtu 16436 qdisc noqueue state UNKNOWN
inet 127.0.0.1/8 brd 127.255.255.255 scope host lo
inet 127.0.0.2/8 brd 127.255.255.255 scope host secondary lo
2: eth0: BROADCAST,MULTICAST,UP,LOWER_UP mtu
One way to do this (there are probably more and better ways). Incoming call
to 123456789 launches meetme(1234,b(connecta.agi))
Connecta.agi calls lines B and C and connects them to meetme(1234).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Tzafrir,
I am in front of the server.
Le dimanche 30 octobre 2011 à 22:13 +0100, Eric van der Vlist a écrit :
Tzafrir,
Le dimanche 30 octobre 2011 à 10:30 +0200, Tzafrir Cohen a écrit :
The problem is elsewhere. What happens if
you manually run:
/usr/share/dahdi/xpp_fxloader load
on 11/01/2011 03:25 PM Danny Nicholas wrote the following:
One way to do this (there are probably more and better ways). Incoming call
to 123456789 launches meetme(1234,b(connecta.agi))
Connecta.agi calls lines B and C and connects them to meetme(1234).
Thanks, but could you be more elaborate
There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin
directory by developing it yourself.
On Tue, Nov 1, 2011 at 6:57 PM, Thanasis thana...@asyr.hopto.org wrote:
on 11/01/2011 03:25 PM Danny Nicholas wrote the following:
One way to do this (there are probably more and
Although if you dig through the archives you can find a good cross-section
of AGI samples. Check the Asterisk Cookbook wikis as well.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind
Sent: Tuesday, November 01, 2011 9:08
You need simple dialplan of four steps:
same =n,Set(conf_name=conf-${RAND(1,1000)})
same =n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =n,MeetMe(${conf_name},dFI1xAC)
same =n,Noop(do post conference stuff)
2011/10/31 Thanasis
Greetings-
I'm about to dive into the process of virtualizing some of my Asterisk
(primarily 1.4.x) infrastructure. In the past, when looking at virt solutions,
the primary issue preventing me from moving was the lack of proper timing. We
do not need it for MeetMe but rather for IAX2 trunking.
on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:
You need simple dialplan of four steps:
same =n,Set(conf_name=conf-${RAND(1,1000)})
same =n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same
Have you thought about using LXC rather than OpenVZ.
There are a few references to allowing guest access to timing hardware online.
I've only been playing with it recently and haven't used it in production yet
but plan to soon.
As for general thoughts about virtualising asterisk, I tried it in
2011-11-01 18:08, Tim Nelson skrev:
Greetings-
I'm about to dive into the process of virtualizing some of my Asterisk
(primarily 1.4.x) infrastructure. In the past, when looking at virt
solutions, the primary issue preventing me from moving was the lack of proper
timing. We do not need it
It would be nice if we can get it going with KVM. Cloud computing solutions
are moving towards the true linux based kernel vs. FreeBSD of XEN.
Cheers,
Nick.
On Tue, Nov 1, 2011 at 5:46 PM, Johan Wilfer li...@jttech.se wrote:
2011-11-01 18:08, Tim Nelson skrev:
Greetings-
I'm about to
Look at upgrading to at least 1.6.2 or 1.8, these both have newer timing
sources that don't rely on dahdi. Also, if conferencing is a big deal, look at
10, this contains a complete rewrite of ConfBridge which doesn't require dahdi
for mixing at all.
Thanks,
--Warren Selby, dCAP
On Nov 1,
Hi,
I have a 1.6.2.6 fax installation with a FFA license which seems to be
installed correctly (in fax show stats, I see that I have 1 Digium
G.711 licensed channel, and 1 Digium T.38 licensed channel).
When trying to call my business line with a fax machine, it looks like
it's ringing to
Type in asterisk CLIcore show application meetme
or google asterisk cmd meetme simple?
On Tue, Nov 1, 2011 at 10:33 PM, Thanasis thana...@asyr.hopto.org wrote:
on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:
You need simple dialplan of four steps:
same
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