Hi all,
Refer to http://www.voip-info.org/wiki/view/Asterisk+config+features.conf.
It shows we can use variable BLINDTRANSFER to call back the one who
transfer the call. However, in my tests below. The result is not as
expected.
case 1:
A calls B (dial(sip/B||Tt)
B answers and connects to A
B
Hello
If we need to save CDRs on different databases for same Asterisk server ie
suppose for context [abcd] save to local:5432:abcd
and for context [wxyz] save to local:5432:wxyz
Can we manage it ? or we need to do some thing in AGI
--
Kind Regards
Shakeel Abbas
Hi,
I'm experiencing frequent kernel panics on a system with Asterisk 1.4.26.3.
There is no core dump, just a kernel panic.
This is the only data I could copy from the screen:
EIP: 0060: [f8e248b4] Tainted: P VLI
EFLAGS: 00210297 (2.6.23-gentoo-r8 #1)
eax: 0130 ebx: ecx: 00220028
Hi,
I had the same problem as yours, and I am using a deadagi which copy the
content of the table CDR is the mine.
I am using the uniqueId value in CDR to know wich row I have to copy.
regards
Mickael
2009/11/18 ABBAS SHAKEEL shakeel.abbas@gmail.com
Hello
If we need to save CDRs on
Ahan thats great thanks..(I was already doing this)
On Wed, Nov 18, 2009 at 3:56 PM, mickael ropars mrop...@gmail.com wrote:
Hi,
I had the same problem as yours, and I am using a deadagi which copy the
content of the table CDR is the mine.
I am using the uniqueId value in CDR to know wich
Hello.
Please help me with this, I can find any solution on this pls help. Your help
will be very appreciated. Thanks.
--- On Tue, 11/17/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: Re: [asterisk-users] can't call through voip provider
What Hardware are you using?
What OS are you running?
If your getting a kernel panic you can install a crashkernel (kdump) and
upon receiving a kernel panic it will reboot to a crashkernel, capture
the crashinfo and safely reboot the system. You can then use the crash
utility to analyse the
On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote:
Please help me with this, I can find any solution on this pls help. Your help
will be very appreciated. Thanks.
It appears that Asterisk keeps sending an SIP INVITE message to your
provider, but not getting any kind of response. After a
Leif Neland schrieb:
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
X-Lite?
http://www.counterpath.com/x-lite.html
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan
Using Asterisk 1.6.1.9, I'm looking for a way to share an extension
between a SIP phone (Cisco 7940) and a SLT on a FXS port of a Cisco 1760
(via sip) -- at any given time I want to be able to pick up either phone
and it should be bridged to the other - just like having two SLTs on the
same
Is it possible to make use of queues for incoming calls but to have
agents that do not need to log in ?
If I create a queue and make certain SIP-users member of the queue, do
these SIP-users always need to log in to the queue to be able to receive
calls that are in the queue ??
Can't a member be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
jonas kellens wrote:
Is it possible to make use of queues for incoming calls but to have
agents that do not need to log in ?
Make the phones members of the queue. In queues.conf:
[MY_QUEUE]
member = SIP/1234
member = SIP/5678
etc.
Barry
Simply use
member=SIP/Tarek
member=IAX2/JONAS
member=LOCAL/whatever
simple and good..
with member=SIP/extension i'm facing a CALL WAITING issue.. the agent hears a
callwaiting signal whenever the queue tries to call .. so i woul dsuggest using
call-limit and busy limite with all your Agents
Hello everybody,
I have a question about value dst of cdr table in asteriskcdrdb, so in
my db I see many registers with letter s in dst field, I found a
opinion that s is the default extension and the all calls enter the
system as s and are then changed as they pass through the dial plan,
so, if
--- On Wed, 11/18/09, Robert Grignon rgrig...@fleetone.com wrote:
What Hardware are you using?
What OS are you running?
If your getting a kernel panic you can install a
crashkernel (kdump) and
upon receiving a kernel panic it will reboot to a
crashkernel, capture
the crashinfo and safely
Hello,
I have an AGI (in C) on 1.4.26.3 that puts a caller on hold, does a few
things, then blind transfers the call (with EXEC Dial...) to a parking
space. This is working fine.
Now I want to add an overhead page AFTER the transfer has happened,
basically announcing that there is a caller
Lookin for anyone who has experienced an issue similar to this. It's quite
baffling as I'm unable to locate much help when it comes to debugging such
an audio oddity.
I'm currently running Asterisk 1.2.18 with a T1/E1 PRI.
To cause the audio bleed (is audio bleed actually what I should even
--- On Wed, 11/18/09, Robert Grignon rgrig...@fleetone.com wrote:
BTW, the last log entry I have in /var/log/asterisk/full
And the last log entries in /var/log/messages are:
Nov 18 10:08:20 voip2 asterisk[25572]: rc_avpair_new: unknown attribute
1490026597
Nov 18 10:08:20 voip2
Title:
Hi everybody
I have a big problem for installing my card
I have the following error but can't find any response on google.
Hope you can help
what is failed with error -5
wctdm24xxp0: BAR0 is not IO Memory.
wctdm24xxp: probe of :00:08.0 failed with error -5
System is debian
At 07:06 AM 11/18/2009, you wrote:
I know that I'm not looking for Dial(SIP/xSIP/y) - as documented, this
handles nothing like what I'm looking for.
It's not the answer you're looking for, but that feature is built
into a Aastra 480i-CT and I think a 57i-CT.
Ira
It's a driver problem. See this link.
http://lists.digium.com/pipermail/dahdi-commits/2009-October/001348.html
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sylvain
MEYNELLY (NEWTEK)
Sent: Wednesday, November 18, 2009
On 11/18/2009 12:08 PM, Sylvain MEYNELLY (NEWTEK) wrote:
I have the following error but can't find any response on google.
Hope you can help
what is failed with error -5
wctdm24xxp0: BAR0 is not IO Memory.
wctdm24xxp: probe of :00:08.0 failed with error -5
The probe failed with error
On 11/16/2009 09:42 AM, Ex Vito wrote:
Shaun,
Thanks for your feedback. See my inline comments.
On Fri, Nov 13, 2009 at 7:18 PM, Shaun Ruffellsruff...@digium.com wrote:
It appears there may be a regression in dahdi-linux 2.2.0 with regards to
the wcte12xp driver and the VPMADT032
Title:
Ok thank you for giving to me a direction
I copy all the driver from a working server with same hardware and
paste into this machine.
Everything working
What I don't is why same distribution and same hardware give me this
problem.
Thank you very much for your help
Bye
--
Hello Xavier,
Unfortunately we are not aware of any Asterisk configuration which will
protect against of a brute force attack on SIP.
We use BFD - http://www.rfxn.com/projects/brute-force-detection/ .
We have found first details here: http://engineertim.com/?cat=15 and we are
currently
Please forgive this off-topic post... I've been on this list since
2005 (over 45k messages in my archive) and this is obviously really
not something I normally do.
If you have a minute and are feeling generous, please visit
http://bailout.chipin.com/ and consider helping me out.
Sorry if I've
Thanks for replying.
But how come I'm able to use a softphone to place calls from withing the lan? I
really dont get it. What ports should I enable in the INPUT chain?
--- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote:
From: Jared Smith jsm...@digium.com
Subject: Re:
What does your provider see when you attempt to call them?
Thanks,
--Warren Selby
On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com
wrote:
Thanks for replying.
But how come I'm able to use a softphone to place calls from withing
the lan? I really dont get it. What ports
According to the provider he says he doesn't see anything coming in on their
side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new
connections. I thought when asterisk starts a communication with a remote
server using an unprivate port to port 5060 theres already an
According to what I know, you have to have 5060 open out and 1-2
open in (you can cut this to as small as 1-10004).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Wednesday,
Danny Nicholas escribió:
Could it be your using option X when you have no extensions for the user to
exit to (therefore when they press dtmf instead of one and done, they just
keep going?)
_
From: asterisk-users-boun...@lists.digium.com
Ok. I do NOT have ports 1-2 opened in. I guess I should try that and
see if it works.
I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I
will keep you posted.
Thanks.
--- On Wed, 11/18/09, Danny Nicholas da...@debsinc.com wrote:
From: Danny Nicholas
Here's an idea - let's say that 4,7 and 8 are extensions that you want to
have a valid action take place on exit from the conference; you already have
that set up in the dialplan. Therefore, you just need to set up
1,2,3,5,6,9,0 and * to do something like either hangup or jump back to the
To be perfectly complete, exactly which inbound ports to open will depend on
the phones in use. For example, a Cisco 7940 (using this example because I
have one on my desk at the moment), the default ports from the config are:
voip_control_port : 5060
start_media_port : 16384
end_media_port :
Philipp Kempgen skrev:
Leif Neland schrieb:
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
X-Lite?
http://www.counterpath.com/x-lite.html
Philipp Kempgen
It does not
subscribe to hints on Asterisk.
Leif
Hi All,
I must say that there are many ways to detect password attack cause this
information actually goes into logs and it's possible to analyze them.
Couple of hours thinking + day or 2 creating gives a really nice result.
Bad thing is that by the time someone will start guessing password with
Anyway to Increase Volume gain in Asterisk ?
USING g729 codec.
___
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Hi,
I am trying to use asterisk open source version(asterisk-1.6.0.5) with
MySQL (using res_odbc)support for extensions and users list.
The call rate is 7 calls / second and each call stays for 120 seconds.
after making 25000 successful calls , calls started
failing with following message on
Hi,
I'm using a revision 6822-enabled Dahdi-Tools (see
https://issues.asterisk.org/view.php?id=13897) with a Junghanns QuadBRI.
1. Do I still need qozap driver ? If positive, how is it recommended to get
it ?
2. Which line should be included in /etc/dahdi/modules to have the
appropriate driver
Hello,
I am using Asterisk 1.4.24.1 version in production.
OS is Centos 5.3 64 bit RAM is 8 GB.
I am facing crash in asterisk approx each 12 hour.
When it crashes I see below linesin asterisk logs.
[Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread
[Nov 18 06:47:23]
Ira skrev:
At 07:06 AM 11/18/2009, you wrote:
I know that I'm not looking for Dial(SIP/xSIP/y) - as documented, this
handles nothing like what I'm looking for.
It's not the answer you're looking for, but that feature is built
into a Aastra 480i-CT and I think a 57i-CT.
Do you
Customer is delivering stuff over the ocean.
Time of delivery is between 1 month to 1.5 months.
So
customers need some sort of a tracking system ( hard to implement given
the conditions ) or he needs to let tjem know when the packages
arrived.
Customers in Europe all have mobile phones, while
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