On Tue, 2012-01-31 at 15:52 -0500, John Knight wrote:
I like the idea of LTR release more often that would have the
feature patches baked in. Case in point the new conference app
requires a jump to version 10 while the 1.8 conference app is quite
useless but 1.8 is my LTR version so I am
* And, is it necessary to use both my server specific certificate and
the intermediate certificate on the telephones or will the telephones
only require the server specific certificate?
The phones should already have the root certificate for Geotrust, you
should not deploy intermediate roots
Thanks for the clarification. I have looked at Polycom's website and
saw which phones have the latest firmware (or at least a firmware that
supports TLS) available.
Didn't get around to the testing with the chained certificate but will
try again this evening.
* And, is it necessary to use
Hi,
I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they'd be diverted to
the extension
M…
-Original Message-
From: Kingsley Tart kings...@skymarket.co.uk
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 01 Feb 2012 10:34:07
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List -
Hi
I have noticed new behaviour of asterisk 10.0 realtime.
In 1.6 when I was using realtime:
[somecontext]
exten = someexten1..
exten = someexten2..
exten = someexten3..
exten = someexten4..
switch = Realtime/${CONTEXT}@extensions
switch statement was executed after
On 01/02/12 10:58, Stuart Elvish wrote:
Thanks for the clarification. I have looked at Polycom's website and
saw which phones have the latest firmware (or at least a firmware that
supports TLS) available.
Didn't get around to the testing with the chained certificate but will
try again
Hello List!
I'm searching for SIP-Providers in the following countries:
Russia
Ukraine
Poland
I need a geographical number for each country, maybe a prepaid
SIP-Account, trunking is not important.
Has anyone some experience with these countries?
yours
christian
--
Can't help you with the SIP account but for geographical numbers in all
3 countries that you mentioned try http://www.globalnumbers.de -
forwarding to any SIP destination is free.
Am 01.02.2012 13:29, schrieb Christian Gansberger:
Hello List!
I'm searching for SIP-Providers in the following
Hello,
I'm using ConfBridge in an application where I need a conference admin
to be able to start and stop recording using a conference menu option.
Currently, I'm doing this by defining ConfBridge menu options
7=dialplan_exec(conference_functions,rec_start,1)
On 02/01/2012 11:42 AM, Josh Freeman wrote:
Hello,
I'm using ConfBridge in an application where I need a conference admin
to be able to start and stop recording using a conference menu option.
Currently, I'm doing this by defining ConfBridge menu options
Hi all,
I'm getting one way audio when calling over the SIP trunk i.e. end device B
(remote end of SIP trunk) can hear device A (softphone registered with
Asterisk) but device A can't hear device B. Even though I configured same
NAT configurations on other servers and they are working good. The
Hi All;
I heard from some friends that there are a dsl router that has Linux OS and it
has asterisk on it, so the ip phone can register on this router, also if the
router has FXS or FXO ports then it can be used to place calls through them.
Is it really? Where I can these routers? Did anyone
On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir ahmedmunir...@gmail.com wrote:
Hi all,
I'm getting one way audio when calling over the SIP trunk i.e. end device
B (remote end of SIP trunk) can hear device A (softphone registered with
Asterisk) but device A can't hear device B. Even though I
At 06:05 AM 1/31/2012, you wrote:
On 01/31/2012 12:17 AM, Ira wrote:
Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk.
On 10.1.0 and trunk, I can't successfully enter the password for any
mailbox in voicemailmain on my Aastra 480i phones. All four version work
with a
Hello,
I have a server that connects to my Voice Server provider so far is working
great! I have a second server that I want to set caller id to a different
number second server I'm going to call it server B. And server B will go
through server A which is connected to my Voice Server Provider.
On 02/01/2012 12:46 PM, Ira wrote:
I notice that comedian mail has instead of [] brackets. Does that
mean it's on its way to being deprecated?
I assume you are referring to how app_voicemail (not 'comedian mail')
is listed the menuselect tool. Umm... no, those are completely
unrelated. How
At 02:31 PM 2/1/2012, you wrote:
app_voicemail (on some systems) requires that res_adsi and res_smdi
be built and loaded; if they are not enabled, then the 'checkbox'
for app_voicemail changes to angle brackets.In menuselect, the
presence of angle brackets instead of square brackets means that
G
Have you ever heard of Google?
Here is a link on google:
http://lmgtfy.com/?q=google
On Wed, Feb 1, 2012 at 2:17 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
I heard from some friends that there are a dsl router that has Linux OS
and it has asterisk on it, so the ip phone can
On 02/01/2012 02:17 PM, bilal ghayyad wrote:
Hi All;
I heard from some friends that there are a dsl router that has Linux OS
and it has asterisk on it, so the ip phone can register on this router,
also if the router has FXS or FXO ports then it can be used to place
calls through them.
Is it
On Wed, Feb 1, 2012 at 5:41 PM, C F shma...@gmail.com wrote:
G
Have you ever heard of Google?
Here is a link on google:
http://lmgtfy.com/?q=google
JAJAJAJAJA!!
--
_
-- Bandwidth and Colocation Provided by
I am trying to configure Asterick, having the following system setup on
the Asterick server:
* eth0 faces the external Internet interface, *but* it does not have IP
address (it has a private one given to it by my ISP's DHCP server);
* eth1 faces my internal network (say 10.1.1.0/24);
* tun0
Greetings,
I currently have an Asterisk 1.8.8.1 system set up with SIP accounts
as well as a Wildcard TDM400P REV I card with both FXS and FXO
ports - FXO is connected to outside lines, FXS connected to inside
analog phones. Everything about the setup works fine except one thing -
after making
Hi all,
I've tried search this problem on the list... no luck...
The case is:
without externip/localnet config on sip.conf [general] my SIP trunk works,
but with no audio NAT problem (asterisk sends the private 192 address to
the outside...)
when I configure externip/localnet correctly my
Whats asterick?
On Wed, Feb 1, 2012 at 7:48 PM, Josh mojo1...@privatedemail.net wrote:
I am trying to configure Asterick, having the following system setup on
the Asterick server:
* eth0 faces the external Internet interface, *but* it does not have IP
address (it has a private one given to
On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour
ortiz.ad...@gmail.com wrote:
Hi all,
I've tried search this problem on the list... no luck...
The case is:
without externip/localnet config on sip.conf [general] my SIP trunk works,
but with no audio NAT problem (asterisk sends the
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