We have a extension pad on our Yealink phone for the receptionist. With our
old non-voip PBX system, the receptionist could pickup a specific extension
by pressing the corresponding key. Is this possible with Asterisk too?
I have configured Asterisk to pickup a specific extension with *59exten.
Hii
Am Montag, den 11.06.2012, 16:12 -0700 schrieb motty.cruz:
Hello,
How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8
exten =
666,1,SIPAddHeader(Alert-Info:http://1.2.3.4/ringtones/ghost.wav)
exten = 666,n,Dial(SIP/10)
The above would not how to
Hi list,
I want to use rtptimeout function on asterisk 1.4 but any docs I read,
it is said that I need to configure it in sip.conf file,
But can I use rtptimeout in users.conf file or do I need to configure
all the SIP accounts on sip.conf file before I can use rtptimeout ?
thanks.
--
If you properly link users.conf to sip.conf you can use it it there too.
2012/6/12 Rabary te...@gulfsat.mg:
Hi list,
I want to use rtptimeout function on asterisk 1.4 but any docs I read, it is
said that I need to configure it in sip.conf file,
But can I use rtptimeout in users.conf file or
This is as easy as running an AGI on your 911 rule to do whatever you
want. The AGI can dial multiple phones, send emails, page you, etc.
Even without the AGI you can do many things from the dialplan.
On Sat, 2012-06-09 at 07:51 -0600, Nunya Biznatch wrote:
Can you set up asterisk so
Thanks for your answer, I will try with kamailio.
2012/6/9 Anton Kvashenkin anton.juga...@gmail.com
Why do you want to use asterisk? It would be better to use kamailio or
opensips for that purpose.
09.06.2012 1:32 пользователь Ignacio Gonzalez mylan...@gmail.com
написал:
Hi everybody, I
On Tue, Jun 12, 2012 at 9:44 AM, Carlos Chavez cur...@telecomabmex.comwrote:
This is as easy as running an AGI on your 911 rule to do whatever
you
want. The AGI can dial multiple phones, send emails, page you, etc.
Even without the AGI you can do many things from the dialplan.
Thank you John. This is a much more elegant solution since I have already
defined 'mailbox' for my SIP device. I'm now using this in my dial plan.
Chet Stevens
asterisk-users@lists.digium.com writes:
Also, related to this question; is there a best or recommended method to
determine the
VMAuthenticate works but I didn't find it to have the type of control that I
was looking for such as using custom prompts separate from the voice mail
prompts, controling how many times the user is prompted, etc. I was able to
come up with the exact
solution I was looking for combined with John
Hello,
I don't know if this list is appropriated to this subject but I want to ask
you if there's some list where I can make an advertising announce for a new
sip web site that was just launched.
hank you.
--
_
-- Bandwidth and
Hi,
I'm just trying to install the DPMA on my Asterisk. I already made the
upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did:
*mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185
*
*compiling Asterisk-Cert2 1.8.11*
*./configure
make
make install
make config
*
Afther
This has come up before on the list and archives but I don't seem to
find a solution for this. On just a few nodes we have this situation
where we see the IP disappear from the CLI iax2 show peers list but
the status shows OK:
3012/3012(Unspecified) (D) 255.255.255.255 0
On 06/12/2012 02:56 PM, Danny Dias wrote:
Hi,
I'm just trying to install the DPMA on my Asterisk. I already made the upgrade
from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did:
/mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185
/
*compiling Asterisk-Cert2
Thanks Jason,
I didn't see in any document with an advice of packages needed. And yes,
i did open a case yesterday, no answer yet!
BR
2012/6/12 Jason Parker jpar...@digium.com
On 06/12/2012 02:56 PM, Danny Dias wrote:
Hi,
I'm just trying to install the DPMA on my Asterisk. I already
It's weird, already installed avahi with yum install avahi
Now the error is:
[Jun 12 16:13:46] WARNING[19949] loader.c: Error loading module
'res_digium_phone.so': /usr/lib/asterisk/modules/res_digium_phone.so:
undefined symbol: ast_vm_msg_play
[Jun 12 16:13:46] WARNING[19949] loader.c: Module
On Tue, Jun 12, 2012 at 10:17:46PM +0200, Danny Dias wrote:
It's weird, already installed avahi with yum install avahi
Now the error is:
[Jun 12 16:13:46] WARNING[19949] loader.c: Error loading module
'res_digium_phone.so': /usr/lib/asterisk/modules/res_digium_phone.so:
undefined symbol:
Hello,
I have an issue I remember seeing a while ago and forgot to investigate
further. Now it is turning into an issue and will need to be resolved. A
customer has Polycom 335 phones (and a couple Soundstation 6000s), and when an
extension is calling out, the screen on the 335 shows the
asterisk biz
On Tue, Jun 12, 2012 at 3:10 PM, Jonson Player jonsonpla...@gmail.com wrote:
Hello,
I don't know if this list is appropriated to this subject but I want to ask
you if there's some list where I can make an advertising announce for a new
sip web site that was just launched.
hank
Thanks for the response. You gave me some ideas I didn't think of such
as sending a text message to the on-call security person's cell phone.
However, while I know I can get the 911 call to call other phones, I
also need location data. I know there are ways to do it, but I don't
have the
On Tue, 12 Jun 2012, Nunya Biznatch wrote:
I also need location data. I know there are ways to do it, but I don't
have the kung-fu for things like databases, and am wondering if there's
something simple in Asterisk like a flat file used to correlate phone
number and location.
1) The
You're absolutely correct. The 911 system maintenance is a PITA.
Our current PBX routes all 911 calls to an box we call the Proctor
Box. That box does a couple things. First, it assigns an ANI to a
number. So for example, 3-digit extensions get a real 10-digit numbers
assigned. It then routes
On Tue, 12 Jun 2012, Nunya Biznatch wrote:
I will look into the local channel variable in the sip.conf. That sounds
promising. If I populate that data, how does it make it to the display of a
phone on campus? I guess that's a piece I haven't either read or been able to
wrap my head around
Dear All,
I am making asterisk report using CDR values given by asterisk.
I have queues which consist of multiple members (extension). Also, an
extension may be in multiple queues. So, I want CDR to record the
name/number of queue from which the call was originated.
E.g.
*Channel*
Hi,
http://www.voip-info.org/wiki/view/Asterisk+log+queue_log
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
Regards,
Zohair Raza
On Wed, Jun 13, 2012 at 7:38 AM, Pratik Shrestha pratik...@gmail.com wrote:
Dear All,
I am making asterisk report using CDR values given by
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