Displaying 20 results from an estimated 30000 matches similar to: "Asterisk AGI issues (at high load)"
2009 Apr 24
2
listen to prompt before bridging call.
Hi,
Can someone please help to resolve the followinng issue:
We would like an asterisk user to call a number and when the called party
picks up the phone, we play a message (press 1 to accept call, 2 to reject
call). Only when the called party presses "1", do we bridge the call and the
two parties can communicate.
What we would like though is that the person who makes the call be able
2010 Mar 30
2
Priority based softhangup
Hi,
Is it possible to softhangup a channel based on priority. I mean I
want to put some calls in higher priority lets say 100. If all
channels are busy and somebody wants to dial an extension with
priority higher than 100. How can softhangup drop a line which has
priority less than 100? I will appreciate your valuable help.
Thanks
Smir
2003 Oct 17
4
Using channel banks
Hello Everyone,
What kind of hardware setup would I need to do if I want a T1 connection to PSTN
and have 48 users in office with analog phones. Will something work if I have a
T410P card in asterisk and have one T1 going to PSTN and other two to a channel
bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks.
Deepak
2004 Feb 22
2
oh323 codec negotiation
Hello
I had this codec negotiation with oh323 call. i used G723 codec and the provider had G729 as first priority. In this situation what ever number i dial i used get "No one there to answer the call". As soon as i changed my codec to G729 the call went through but had other problems, which i got away by dowloading the latest code for oh323.
Has anyone seen this problem? or it is
2009 Sep 29
2
play audio file within an active call
Hi,
I'm wondering if someone can share their thoughts on how to implement a system that periodically checks active channels which have been up for more than X minutes and plays/injects a sound file. The idea is to simply warn users that they've been on the phone for quite a while and maybe they should consider hanging up. If the call stays up for more than Y minutes, it is dropped
2006 Jun 09
1
hangup extension
I've been testing the debug version of AstTAPI, which worked for a few
calls, then a bit later in the day (and ever since), when the call is
hung up, the TAPI client doesn't get notified.
Looking at the server logs, The TAPI message that is sent upon hangup,
isn't being sent.
exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE)
This is in the same context as
2004 Aug 06
2
Asterisk not starting
Hello!
Asterisk "CVS-HEAD-08/06/04-14:55:13" won't start on two of three different
Gentoo machines. This is the output of gdb:
ultra asterisk # gdb /usr/sbin/asterisk
GNU gdb 6.0
Copyright 2003 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
2009 Feb 19
3
DTMF
IVR Number :17275691533
When I try it from xlite configuring my provider directly, it works
perfectly.
When I try to dial out from dialer , it doesnt work.
[sip8]
type=peer
username=user
fromuser=user
authuser=user
secret=password
host=8.14.146.111
nat=no
canreinvite=yes
insecure=very
disallow=all
allow=g729
allow=ulaw
context=default
dtmfmode=rfc2833
What cld be the reason ?
--------------
2004 Dec 21
10
Codec Selection
Hi,
I have 2 g729 licences - what I want to do is use g729 by default but if
I get more than 2 calls at a time, use gsm for the others.
So, I put this on all my sip providers:
disallow=all
allow=g729
allow=gsm
However, this just seems to use gsm for everything. If I comment out the
gsm line, it then uses g729.
I thought it would use the codec's in the order they are allowed - is
this
2008 Dec 05
2
async agi question
Hi,
I am developing asterisk support for our application using the Async AGI
and Asterisk-Java.
One thing I haven't been able to implement is how to stop playing a
sound. Something similar to StopIO for Dialogic GlobalCall or
DivaStopSending for Eicon.
Is there any way to achieve this today which I have missed? Or could
someone give me hints on how I could implement this in the res_agi.c The
2005 Jun 27
3
Bad Bad Performance; Max 20 Calls on Quad Proc?
Here is the setup: Dell 6250, Quad Proc P3 500Mhz. Digium Single Span T1
card. System has 71 sip peers/users. All calls are G729; we have 10
licenses. All calls follow this path: UA -> Asterisk -> Digium PRI ->
Class4/5 switch. The switch dictates if it should go out local PRI for
local termination or out PRI to our Cisco AS5300 for LD termination.
Our biggest problem is echo.
2005 Jul 20
1
Zap channel(s), meetme and codecs/licences
Hi all,
Some simple questions about codecs:
What codec does the Zap channel use by default?
Can this default be changed, and to what? (g729 too?)
What codec does meetme use? (I think this is ulaw, but asking to be sure)
Can you use another codec, or does everything have to be transcoded to ulaw?
Finally ... if I have a 3way call going, between 1 g729 caller and two
other callers, do I need one
2006 Jan 27
5
External IAX2 phone defined as internal behaving as from PSTN
Have asterisk@home 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls from 1055 get picked up as if it were an external call
(see below) and goes straight to the ring
2006 Jan 29
4
Asterisk + XEN does it make sense?
Hi List,
I was wondering if anybody had tried running Asterisk inside
virtualization software such as Xen. Are there known problems doing it?
Cheers,
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
2005 Feb 24
1
Which Codec(s) to use..?
Hey Everyone,
I am playing around with my * box, and I have a few different phones
hanging off it it right now.
I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco
ATA186 with a Panasonic cordless phone attached to it, I have a Digum
IAXy with a dumb analog phone attached to it, and I have a Linksys
PAP2-NA with an AT&T 959 analog phone attached to it.
I also have several
2007 Aug 27
3
voip provider settings problem, please help
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk
1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my
problem is the following one:
when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown
in the asterisk CLI and caller and callee can hear each other when
2006 Apr 12
33
DUNDi with SIP
Anyone out there have a functional DUNDi configuration using SIP for the
inter-Asterisk transport? I've gotten it to work with IAX2, but if I
change it to SIP it does not pass the call over even though it knows
where to send it. Thanks.
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2003 Nov 21
9
Outline For Asterisk Book - Please Review & Comment
Asterisk Users
In an attempt to help Asterisk move forward, a number of us have decided
to create a book. It would initially be released as an "ebook" that
could be sent to newbies to help them up the rather steep learning
curve. Ultimately I would like to see it published and sold in
bookstores (preferably by O'Reilly & Co.).
Below is the outline for the book. We REALLY
2004 Dec 17
6
Realtime and PostgreSQL
Has anyone had any luck with PostgreSQL and Realtime? The realtime
instructions on voip-info seem pretty straight forward... just not woking for
me. I've included all of my config files below, and my console output.
Entire console bootup output:
[root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing
2004 Apr 01
4
sip problems
chan_sip.c6524 reload_config= unable to get ip address from asterisk,
sip disabled
The ip address is working fine, Internet works great. Can anyone
help...Thanks