On 5/6/11 11:52 PM, Andrew Latham wrote:
On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanianva...@arminco.com wrote:
Has anyone used this board as an Asterisk server?
http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y
I'm mostly interested about the possible
On 11-05-06 02:56 PM, Watkins, Bradley wrote:
Yes, use the MinivmMWI application.
That's how I've done it in the past as well.
Leif.
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New to Asterisk?
Jerry Geis wrote:
Is there a way in asterisk to Activate/Clear the blinking light on
polycom phones
I've always just dumped a msg000.txt in the voicemail directory of that
phone and removed it when not needed. Under 1.4, the Polycoms act on it.
Doug
--
Ben Franklin quote:
Those who
On Sat, May 7, 2011 at 3:05 AM, Vahan Yerkanian va...@arminco.com wrote:
On 5/6/11 11:52 PM, Andrew Latham wrote:
On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanianva...@arminco.com wrote:
Has anyone used this board as an Asterisk server?
On Fri, May 6, 2011 at 2:52 PM, Andrew Latham lath...@gmail.com wrote:
On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote:
Has anyone used this board as an Asterisk server?
http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y
I'm mostly
On Sat, May 7, 2011 at 10:12 AM, Ryan Wagoner rswago...@gmail.com wrote:
On Fri, May 6, 2011 at 2:52 PM, Andrew Latham lath...@gmail.com wrote:
On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote:
Has anyone used this board as an Asterisk server?
Dear;
Where I can find a new documentation for Asterisk 1.8?
Where is the wrong in that line? I see it is as 1.8 version !
500 = 1234,Operator,opera...@gama.com
Regards
Bilal
---
You are using an old format for specifying the
mailbox. See core show application voicemail
- Original Message -
Dear;
Where I can find a new documentation for Asterisk 1.8?
Where is the wrong in that line? I see it is as 1.8 version !
500 = 1234,Operator,opera...@gama.com
Regards
Bilal
---
You are using an old format for specifying the
I know a lot has changed over the past couple of years, and even monthly, and
that Asterisk running within a virtualised environment is very happy indeed. If
one would only be using SIP/IAX would Xen/KVM be the best solution ? / or
perhaps VServer/LXC maybe advantageous due to binary hashing.
Hello List,
i need to be able to record the call transferred from iax extension to sip
extension
when i call the sip extension from the IAX extension i can record the call
without any issue
but when i receive a call from customer in IAX and i transfer this call to
SIP client
the conversation
On Sat, 7 May 2011, salaheddine elharit wrote:
Hello List,
i need to be able to record the call transferred from iax extension to sip
extension
when i call the sip extension from the IAX extension i can record the call
without any issue
but when i receive a call from customer in IAX and i
Dear;
In the extensions. conf, I have the following:
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}@Internal)
So, I am writing the arguements of the Voicemail ( ) wrong?
Regards
Bilal
Dear;
Where I can find a new documentation for Asterisk
1.8?
Where is the wrong in that line? I
Hello all,
I have installed the .deb packages of the Asterisk v1.8.3.3 from the
upstream project on my Debian GNU/Linux Squeeze server and bought the
Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
exercise. After setting up everything and trying to fix this problem,
I am still
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