It's look like I'm wrong, didn't read your reply first, don't know there's such
feature, very nice info :D
--
Best regards,
Rudi
-Original Message-
From: SamyGo
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 7 Aug 2012 09:24:19
To: Asterisk Users Mailing List - Non-Commerci
www.voip-info.org/wiki/view/Asterisk+multi-language
--
Best regards,
Rudi
-Original Message-
From: bilal ghayyad
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 6 Aug 2012 15:43:24
To:
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [aster
AFAIK there's no such feature on IP Phone to retrieve Caller ID before dialing,
because Caller ID on IP Phones is manually set on the Phone, the best thing
that you can do is manually match the destination Phone's Caller ID in the
source Phone's Address Book
--
Best regards,
Rudi
-Origina
Hi,
You can try with SIPCHANINFO function otherwise you need to modify
chan_sip.c for getting this addresses.
thanks
Dhaval
On Tue, Aug 7, 2012 at 12:10 AM, CB wrote:
> We are looking to further secure our Asterisk installation by inspecting
> the
> IP address that a SIP INVITE comes from and
Hi,
You need to set rpid on the calling phone settings, if that phone knows
what to do with RPID. Then you need to set allowrpid=yes in the sip peer
settings of A party and B party. I did that on CISCO 79X0 phones and it
worked perfectly,
Regards,
Sammy
On Tue, Aug 7, 2012 at 3:43 AM, bilal gha
Hi All;
Asterisk 1.8.11-cert1
I need to do the following, how?
If my extension is 500 and I need to call the extension 501, so when dialing
501, then I need to be able to see the name of the 501 (for example, the name
was: Mike, so I need to see at my IP Phone that I am calling Mike which is t
Dears;
I discover that I have to place the wave files in the
/var/lib/asterisk/sounds/custom/
So, can I understand that the only solution I have is to copy the files that
are existed in the path /var/lib/asterisk/sounds/en/ to the path
/var/lib/asterisk/sounds/custom? Or there is any other sol
We are looking to further secure our Asterisk installation by inspecting the
IP address that a SIP INVITE comes from and performing some logic to
determine whether the call should proceed. The purpose of this is to prevent
calls to certain expensive destinations if the SIP message is coming from a
Hello,
Using asterisk 1.6 as sip client to register with sip provider and
terminate calls through them. SIP Provider has provided sip proxy and sip
server details. The problem is that the sip server FQDN does not resolve
on the internet. So I can only presume that the SIP proxy knows how to
rea
I have bought a new server today:
i7-2600 CPU, 8GB and 2 x 256GB SSDs. 100Mbit Connection.
I hope CPU is powerful enough for 200 concurrent calls.
On Sun, Aug 5, 2012 at 1:57 AM, Michelle Dupuis wrote:
> That's how we do it - write to a memory based (ramdisk) disk then write
> to HDD upon c
Thats a great tutorial with very good conceptual details like SIP messages
flow.
Thanks Daniel :)
On Mon, Aug 6, 2012 at 6:48 PM, Daniel-Constantin Mierla
wrote:
> Hello,
>
> I released an update to my series of Kamailio and Asterisk Realtime
> Integration, using the latest stable versions of the
On Mon, Aug 06, 2012 at 10:03:41AM +0200, Giuseppe Longo wrote:
> Hello guys,
> i've a little question to ask. What is the file asterisk.ctl ?
That is a UNIX Domain Socket file used to pass commands to an
Asterisk process. It's how "asterisk -r" and "asterisk -rx"
communicate with the back-end pro
Hello,
I released an update to my series of Kamailio and Asterisk Realtime
Integration, using the latest stable versions of the two projects,
respectively 3.3.1 and 10.7.0. You can find it at:
* http://asipto.com/u/68
The tutorial focuses on how to use Asterisk's database structure to
per
On 06/08/12 02:59, Vladimir Mikhelson wrote:
> Have you tried 1.8.15?
I'm trying 1.8.13 because that is the versions currently scheduled for
release in Debian 7 (wheezy)
http://packages.debian.org/wheezy/asterisk
If 1.8.15 contains definite solutions for TLS problems, then either
a) they can
Hi Paolo,
I had yesterday a similar problem and it was caused by a misconfigured
IP address in extensions.conf that I forgot to update after changing
some IP addresses in my network.
Check the network connectivity between you Asterisk host and "1000".
Double check that the IP address is correct.
Hi all,
question about register refresh time.
One of our supplier had a maintenance work on sat 4 Aug which was
replacing the production server for an Asterisk 1.4 running version.
We have few Asterisk connected to them (version 1.6, 1.8 and 1.10) with
register Username and Passwd. After the
Hello guys,
i've a little question to ask. What is the file asterisk.ctl ?
Thanks,
Regards.
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