Hi all!
I would like to ask, how you realize the following in a call queue:
When a caller gets into a queue how can I play a welcome Message to this caller
first, before he starts hearing the music?
We now use a playback before the caller gets into the queue. But when the queue
is closed
Not true. When you register the /1234 on the end of the line sends it
to that extension in the context you specified in the peer entry with
context=.
Thanks for your help..
Can you please re-explain maybe another way?
I'm having trouble understanding what you mean by:
context you specified
Hi all,
How can i use different VLANs for signaling and audio, e.g vlan 100 for sip
and vlan 200 for rtp? Where can i find documentations for this?
Comments and suggestions are welcomed (a sample config too :-)))
thx in advance
rich
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Hi!
Coco Richard wrote:
How can i use different VLANs for signaling and audio, e.g vlan 100 for sip
and vlan 200 for rtp? Where can i find documentations for this?
Comments and suggestions are welcomed (a sample config too :-)))
If you need to make RTP traffic priority higher then
Steve Totaro wrote:
Just out of curiosity, why did you feel they needed an upgrade?
Thanks,
Steve
For me it was a bit of slapdash behaviour on my part.
I'd hreceived a few complaints that sounded like they were probably
asterisk' fault (most interesting one being that one person could
Hi Antonio,
here is the corresponding section of my jabber.conf file, that allows
our Asterisk server to connect to our local XMPP server (jabberd2), as
a component.
[asterisk-component]
type=component
serverhost=jabber.inria.fr
username=asterisk
secret=***
port=5347
Depending on your XMPP
Hi All;
I do not know why this happen, if anyone faced it and knows any idea about it:
I have analoge phone connected to FXS port, when we dial its extension, the
phone ring first ring, second ring, then it automatically answer the channel
and hangup (as I see in the CLI), anyone can advise
Interesting use of IVR
http://www.mobilemarketer.com/cms/sectors/media-publishing/1233.html
They still have to have the overhead of the premium sms service (trust
me it's an 'overhead', carriers take a ton of money for this service -
check out more info on CSC's in the USA here
I would like extensions within a specific group to capture calls (with the
typical *8).
All's fine if the extensions are registered to the same asterisk server.
However, I don't know how to make it work if they register to different
Asterisk servers within the same LAN.
I use DUNDi to call
Jeff Peeler wrote:
On Fri, 2008-06-27 at 11:36 -0300, Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.13 as our SIP server and several SIP
clients using ZRTP support (with Zfone module in Windows and libzrtp in
Linux).
People say that it's necessary to use an Asterisk patch
On 28 Jun 2008, at 18:36, Tzafrir Cohen wrote:
On Sat, Jun 28, 2008 at 12:16:53PM -0400, Steve Totaro wrote:
On Sat, Jun 28, 2008 at 12:06 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Sat, Jun 28, 2008 at 11:25:44AM -0400, Steve Totaro wrote:
On Sat, Jun 28, 2008 at 10:11 AM, Tzafrir Cohen
the only suggestion i would have for you is to use a SINGLE file for your MOH
.. and you record the welcoming note in the begining of the file.. so whenever
a caller comes in .. they will hear the MOH .. which has the welcoming note
before the music starts...
i know it's a stupid trick but it
Philippe,
Is it possible to configure Asterisk in a component configuration such
that any sip user could call any Jabber user? A sip user could dial
sip:[EMAIL PROTECTED]
, for example, and the call would get routed to that person, even if
the two parties have never spoke before and are
i have been through this with Dynamic agents and callback .. i used to use
addQueueMember but it caused me troubles when joining a queue.. because
sometimes the agent would be in the queue twice..
my suggestion to you is to check if you can make calls between two members of
the queue .. then
[zaptel]
span=1,0,0,esf,b8zs
At least one of your spans should be getting it's timing from your
service provider. It looks like that would be span one, this should
read:
span=1,1,0,esf,b8zs
I checked my config files from before my upgrade, and I do have span 1
setup as you
Does anyone know of a spam filter that will work with Asterisk?
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asterisk-users mailing
Hello, yesterday one of the extensions on my asterisk server got
compromised by brute-force attack. The attacker used it to try pull an
identity theft scam playing a recording from a bank your account has
been blocked due to unusual activity, please call this number...
Attacker managed to make
On Mon, 2008-06-30 at 12:03 -0400, Andrew Joakimsen wrote:
Does anyone know of a spam filter that will work with Asterisk?
What does spam have to do with Asterisk? Or do you mean spit perhaps?
http://en.wikipedia.org/wiki/VoIP_spam ? Probably the same techniques
such as whilelisting,
On Mon, 2008-06-30 at 11:15 -0500, spectro wrote:
I need a way to block that IP from connecting to my
asterisk server, please advice.
netfilter. aka iptables.
b.
signature.asc
Description: This is a digitally signed message part
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Hi Daniel,
I'm intrigued by this and wanted to try it out - but I'm wondering how
you get Asterisk to call sox at all during Voicemail()? Our server
doesn't even have sox installed, so I'm not sure how to go about
tricking Asterisk into running a different one.
To do anything useful you would
The Asterisk.org development team has released Asterisk version 1.4.21.1.
This release includes a critical bug fix for 1.4.21. All users that
experienced lockups when upgrading to 1.4.21 should have their issues
resolved with this update.
Asterisk 1.4.21.1 is available for download from the
Doesn't have to be Voip-originated.. One guy out of his apartment
in Texas constantly orders phone lines, disconnects them after a few
days of continuous dialing. http://customcampaigns.net/political.html
He is not calling with a political message but instead simply
harvesting phone numbers.
hi,
I have an AGI running after an outgoing call file starts it up.
Everything works fine except if my line has a problem.
Trying to simulate this I unplug the line. So there is no dialtone.
How do I detect this and let the AGI know so I can try line 2, 3, 4 etc...
Detecting the the AGI or some
Do a reverse lookup on your attacker.
Then find their ISP.
Then file an abuse complaint.
On Mon, Jun 30, 2008 at 12:15 PM, spectro [EMAIL PROTECTED] wrote:
Hello, yesterday one of the extensions on my asterisk server got
compromised by brute-force attack. The attacker used it to try pull an
On Mon, 30 Jun 2008, Daniel Hazelbaker wrote:
I'm intrigued by this and wanted to try it out - but I'm wondering how
you get Asterisk to call sox at all during Voicemail()? Our server
doesn't even have sox installed, so I'm not sure how to go about
tricking Asterisk into running a
Hi Atis and friends,
I think it has direct relation with calls not being delivered to available
agents, as you can see at following lines, I´ve confirmed i*n situ* there
were real available agents:
[EMAIL PROTECTED] apps]# asterisk -rx queue show
QUEUE1 has 0 calls (max 100) in 'rrmemory'
On Mon, Jun 30, 2008 at 1:31 PM, David Backeberg [EMAIL PROTECTED] wrote:
Do a reverse lookup on your attacker.
Then find their ISP.
Then file an abuse complaint.
already done, also filed a report with FBI cybercrime unit and setup
iptables to block incoming traffic from that IP.
My question
On Mon, Jun 30, 2008 at 11:32:58AM -0700, Steve Edwards wrote:
To do anything useful you would have to get sox installed on your
server.
If you are using RedHat or CentOS, installing sox should be as simple as:
sudo yum install sox
Other distros have similar commands.
One
I forgot it!, I'm using Asterisk 1.4.19.1 version.
On Mon, Jun 30, 2008 at 1:47 PM, Chento Arohuanca [EMAIL PROTECTED]
wrote:
Hi Atis and friends,
I think it has direct relation with calls not being delivered to available
agents, as you can see at following lines, I´ve confirmed i*n situ*
iptables -A INPUT -p tcp -s 74.52.112.162 -j DROP
Good luck.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of spectro
Sent: June 30, 2008 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip extension
better drop udp also.
Mark Hamilton wrote:
iptables -A INPUT -p tcp -s 74.52.112.162 -j DROP
Good luck.
Via: SIP/2.0/UDP
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I just bought a HTC TyTn II phone, but unfortunately it doesn't even have
a SIP client in it.
I tried the wiki searching for a SIP or IAX client but only found some
PocketPC stuff (Windows Mobile 2003).
Does anyone know of a good quality SIP or IAX softphone that will run on
Windows Mobile 6?
You can use a hashtable to watch incoming traffic, sort it into
buckets based on its ip address, and take action accordingly. But
you'll need some method of sorting out legitimate traffic versus bad
traffic. You'll need to come up with some more characteristics than
just that something is
Coco Richard [EMAIL PROTECTED] writes:
Hi all,
How can i use different VLANs for signaling and audio, e.g vlan 100 for
sip and vlan 200 for rtp? Where can i find documentations for this?
That would be very difficult. You can do it in Linux with firewall
rules and policy routing based on
Someone should write an asterisk-centric document on this topic, it's
likely to become an issue someday. Sounds like a great subject for
VoIP USers Conference as well. Any volunteers?
/r
ssh hack detection is easy because each new bruteforce starts with a
tcp syn, so you can count them and
Hi Remco,
Both of these may be helpful to you, one to fix the SMS issue, and one to
enable the stock MS voip client:
http://www.mattgibson.ca/2008/04/13/fix-sms-time-issues-on-rogersfido-unlock
ed-gsm3g-windows-mobile-56-smartphonespocketpcs/
and
On 6/30/08, randulo [EMAIL PROTECTED] wrote:
Someone should write an asterisk-centric document on this topic, it's
likely to become an issue someday. Sounds like a great subject for
VoIP USers Conference as well. Any volunteers?
iptables string and limit matching could be a start, although
On Mon, Jun 30, 2008 at 5:10 PM, Kristian Kielhofner
[EMAIL PROTECTED] wrote:
Does anyone want to write a kernel module? ;)
The thing I was mentioning about hashing addresses is already in the
kernel, check out:
hashlimit on google,
or net/netfilter/xt_hashlimit.c in your favorite 2.6 kernel
Try some of the shell scripts in the asteriskcookbook recipe heap
http://asteriskcookbook.com/wiki/index.php/RecipeHeap
Specifically
http://asteriskcookbook.com/wiki/index.php/Asterisk_Brute_Force_Prevention
Cheers Duncan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
Hi,
A couple of our users are reporting that intermittently, their
voicemails are unable to be heard because there is a
milliwatt-sounding tone recorded over the top of it.
Has anyone else encountered this issue?
I have put a recording of the voicemail up online for people to listen
to to see what
On 6/30/08, David Backeberg [EMAIL PROTECTED] wrote:
The thing I was mentioning about hashing addresses is already in the
kernel, check out:
hashlimit on google,
or net/netfilter/xt_hashlimit.c in your favorite 2.6 kernel source
The other cases you mention could be done with multiple
Hi,
After doing a yum update on my previously Centos-5.1 system, now
zaptel-1.4.11 fails to build with this below.
I figured I better caution people because it sucks being down, and
I haven't figured out how to install the previously compiled modules
into the new kernel or any other work
Hi All;
I enabled the dnsmgr.conf and I put the refresh to be 300, but I am not able to
see the sentence in the CLI:
Refreshing DNS lookups
But I see this sentence on other asterisk machines.
What could be the reason?
Regards
Bilal
___
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On Jun 30, 2008, at 1:04 PM, [EMAIL PROTECTED]
wrote:
But to get asterisk to run a different/fake sox, just install
whatever
you want to run as /usr/local/bin/sox and then edit your
safe_asterisk
script as I mentioned below.
I think this is a bad approach. It's going to be a big gotcha
Well, that was sorta my point.
CP
Steve Edwards wrote:
A quick grep through the Asterisk (1.2.28) sources shows res_monitor using
soxmix if the channel variable MONITOR_EXEC is not defined -- but nothing
in app_voicemail. Am I missing something?
Greetings..
i have 20 extensions with two queues.. i have members in the queues as SIP/
now recently i have noticed that users are unable to call each other.. this is
causing me a headache..
calls comming to the queues are forwarded smoothly to the users.. but they
can't call eachother..
On June 30, 2008 06:25:17 pm Mark G. Thomas wrote:
Hi,
After doing a yum update on my previously Centos-5.1 system, now
zaptel-1.4.11 fails to build with this below.
CC [M] /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o
In file included from
On Mon, Jun 30, 2008 at 03:47:48PM -0400, Matt Watson wrote:
On June 30, 2008 06:25:17 pm Mark G. Thomas wrote:
Hi,
After doing a yum update on my previously Centos-5.1 system, now
zaptel-1.4.11 fails to build with this below.
CC [M]
Matt Watson wrote:
On June 30, 2008 06:25:17 pm Mark G. Thomas wrote:
Hi,
After doing a yum update on my previously Centos-5.1 system, now
zaptel-1.4.11 fails to build with this below.
CC [M] /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o
In file included from
Hi There,
I am looking to build an Asterisk server with dual AVM Fritz!PCI cards
linked to 2 BRI in New Zealand. Just wondering if anyone has done
this, and if you have any ideas about the best disto choice for this
task?
Simon
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On June 30, 2008 08:44:44 pm Simon wrote:
Hi There,
I am looking to build an Asterisk server with dual AVM Fritz!PCI cards
linked to 2 BRI in New Zealand. Just wondering if anyone has done
this, and if you have any ideas about the best disto choice for this
task?
Let me be the first to
Hi guys,
I'm recording all the calls ingressing to a queue using FreePBX, but the
output file is, for example, this:
q501-20080601-072010-1212322768.57.wav
Where 501 is the queue name, 20080601 is year+month+date, 072010 is the
hour, and 1212322768 is date+hour in unixtime format.
I want to
Matt Watson wrote:
On June 30, 2008 08:44:44 pm Simon wrote:
Hi There,
I am looking to build an Asterisk server with dual AVM Fritz!PCI cards
linked to 2 BRI in New Zealand. Just wondering if anyone has done
this, and if you have any ideas about the best disto choice for this
task?
Let
Hello Tarek,
thank you for your idea. But this only would work for the first caller - when
the moh starts.
all other callers go directly into moh on the position where the first caller
is in moh.
So this does not work. :-(
Anyone an other idea?
thank you
Martin
- Original
I am after someone to help me to config H323 on asterisk if possible since I
am far too busy stuck on another project. Interested parties please msn me
on sam _ _ tam AT hotmail.com please take out all space and change AT to @
If you are unsure then you can always email me with your contact via
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