Tilghman Lesher tilgh...@meg.abyt.es writes:
Correct; and Asterisk needs to be started as root, even if it will drop
privileges after startup. Do this, and there should be no problems.
Starting as root + dropping privileges is fine. Running configure as
root is not so fine; that basically
Hallo everybody,
I got a question to asterisk 1.6. Is it possible to playback a Audiofile in
uplink and to record the downlink channel in another Audifile at the same
time?
If it is possible, how should I do it? Please explain it.
Thank you for your help to my thesis!
best regards,
Felix
--
Dear Mr/Ms;
web have some Queues and our Call Center and put caller in Queue Based on
some regional decisions.
by the way, after the Caller placed on Queues, we like to be able to reorder
them on our rules.
as an example:
there is a queue which have 10 caller in waiting stage right now, one with
exten = _9944NX,1,Answer()
exten = _9944NX,2,Noop(GOING FOR THE AGI)
exten = _9944NX,3,Noop(XX)
exten = _9944NX,4,Noop()
exten = _9944NX,5,AGI(//Some script here it works perfectly fine)
exten = _9944NX,6,Noop(AGI
On 11-02-01 04:02 AM, Felix Dong wrote:
I got a question to asterisk 1.6. Is it possible to playback a Audiofile in
uplink and to record the downlink channel in another Audifile at the same
time?
Yes, look at MixMonitor.
*CLI core show application MixMonitor
--
Paul Belanger
Digium, Inc. |
Hello list,
what musiconhold class has priority :
- field musiconhold of the SIPaccount and field musiconhold of a queue
or
- Set(CHANNEL(musicclass)=)
??
Kind regards,
Jonas.
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, February 01, 2011 10:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Musiconhold priority
Hello list,
what
Is it possible to SIP trunk to this Cisco device so that phones connected to
the Cisco box can dial extensions on the Asterisk box?
What I would like to be able to do is dial some extension(s) on phones
connected to the Cisco box and have the call routed into extension(s) on the
Asterisk box.
Hello,
I've defined some new musiconhold classes in musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
;
[908001]
mode=files
directory=/var/lib/asterisk/moh/908001
random=yes
;
[101001-1]
mode=files
directory=/var/lib/asterisk/moh/101001/1
random=yes
;
[101001-2]
How can I use the application Monitor() in the Python AGI skripts?
Thanks a lot.
best regards,
Feilx
--
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New to Asterisk? Join us for a live introductory
Yes, you can use the Mixmonitor command.
But if you want to have only one party on the recording, you should use the
Monitor command without the 'm' option.
http://www.astblog.com/2011/02/01/asterisk-mixmonitor-cmd/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, February 01, 2011 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to load new musiconhold classes ?
Hi again,
Nobody knows how to disable it? Can at least someone pinpoint me where
can I check the latest documentation regarding SRTP. Maybe something
might have change in the meanwhile 'Cause so far it looks like there is
a bug in asterisk.
Well, maybe I should report this bug then.
- Miguel
I am as well
- Original Message -
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue Feb 01 11:22:41 2011
Subject: Re: [asterisk-users] How to
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel
Baptista
Sent: Tuesday, February 01, 2011 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to disable srtp in asterisk
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel
Baptista
Sent: Tuesday, February 01, 2011 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
2011/1/24 Matt Riddell li...@venturevoip.com
Hi all,
So, we reverted the LibPRI version and tested it, and then tried with the
latest version of everything. Still no changes.
The BRI line is in PTMP. If we set the configs to PTMP in the
genconf_parameters and try it, we get the
On Tuesday 01 February 2011 02:24:20 Benny Amorsen wrote:
Tilghman Lesher tilgh...@meg.abyt.es writes:
Correct; and Asterisk needs to be started as root, even if it will
drop privileges after startup. Do this, and there should be no
problems.
Starting as root + dropping privileges is
According to chapter 7 (Outside Connectivity) of the excellent Asterisk:
The Definitive Guide (review version online at
http://ofps.oreilly.com/titles/9780596517342/index.html), the following
enables secure signaling and media paths:
exten = 1234,1,Set(CHANNEL(secure_bridge_signaling)=1)
same
On Tuesday 01 February 2011 11:49:51 Paul Belanger wrote:
On 11-01-26 02:59 PM, Tilghman Lesher wrote:
On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote:
[CREATECALL]
dsn=Example
writesql=INSERT INTO x (y) VALUES (z)
readsql=SELECT LAST_INSERT_ID();
That assumes you have
Paul Belanger wrote:
On 11-01-26 02:59 PM, Tilghman Lesher wrote:
On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote:
[CREATECALL]
dsn=Example
writesql=INSERT INTO x (y) VALUES (z)
readsql=SELECT LAST_INSERT_ID();
That assumes you have only one call in existence at a time. If two
On Tue, Feb 1, 2011 at 12:30 PM, Danny Nicholas da...@debsinc.com wrote:
Now that my “smart” answer is out of the way, did you try
- srtpcapable=no
- in sip.conf?
reference: http://www.voip-info.org/wiki/view/Asterisk+SRTP
I've been looking at the trunk (1.8.+) code
On Tuesday 01 February 2011 12:36:46 Jose P. Espinal wrote:
Paul Belanger wrote:
On 11-01-26 02:59 PM, Tilghman Lesher wrote:
On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote:
[CREATECALL]
dsn=Example
writesql=INSERT INTO x (y) VALUES (z)
readsql=SELECT LAST_INSERT_ID();
On 11-02-01 01:21 PM, Tilghman Lesher wrote:
Assuming you were using a MySQL backend that supported transactions,
you could use the transaction layer in Asterisk 1.6.2 and greater to ensure
that each channel got a serialized view. That would make this approach
work.
Ya, I think I'm going to
On 02/01/2011 12:34 PM, Harel Cohen wrote:
As one with theoretical knowledge in programing, but never on Linux, I
can understand terms and code structure but I don’t know:
1. What shell commands (e.g. ./configure, make, make install etc.)
should I run to recompile Asterisk (same version)?
On Tue, 1 Feb 2011, Felix Dong wrote:
How can I use the application Monitor() in the Python AGI skripts?
Use the exec AGI command.
I use C so it looks something like this:
exec_agi(exec MONITOR wav|%s/%02d-prompt|m
, recording_path
, idx
Hi Asterisk Users,
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz
On 11-02-01 05:22 PM, Juan David Diaz wrote:
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650, 2.66Ghz, 12M
Tuesday, February 1, 2011, 11:22:30 PM, Juan wrote:
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650,
On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas da...@debsinc.com wrote:
Not sure how queues factor into this equation; guess that’s a “try and see”
thing.
From my experience, the explicitly defined Set(CHANNEL(musicclass)=blah)
takes precedence over a queue's defined moh class.
--
Thanks,
That's quite possible. We handle around 100 similtaneous calls(PRI +
SIP) with a decent dell server with only 4gb ram.
On Wed, Feb 2, 2011 at 6:22 AM, Juan David Diaz juanch...@gmail.com wrote:
Hi Asterisk Users,
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI
Hi everyone
How can I get the current calls details in asterisk.if I use cli
commad core show channels,there is two channels of each call.But the
requirement is, need to get caller ,calee,starttime ,duration of the
current calls.This value should be proper for call forward,call transfer
Hi all,
My experiment scenario is like this:
SIPp Uac - ASTERISK
SERVER--SIPp uas
1. when i had registered bob with this command ./sipp -sf
register_client.xml -inf register1.csv -i 192.168.1.6:5060 192.168.1.6 -p
5061
Hey guys I was hoping I could get a few pointers on a problem I have been
trying to debug for the last couple of months regarding asterisk AGI scripts
and unexpected termination.
I have this agi script that accepts incoming faxes using RxFax on the latest
asterisk 1.4 branch. Its written with perl
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