On Friday 08 June 2012, Steve Sokol wrote:
Allison Smith has kindly agreed to add some new prompts to the additional
prompt set for Asterisk. If you have ideas for additional stock prompts
(serious or silly), please submit them:
Blake Burgess wrote:
We had a bunch of voice quaity issues which took ages to diagnose because
of this. Obviously if you have a DAHDI card that your passing through to
the vm or one of thesehttp://wiki.sangoma.com/sangoma-wanpipe-voicetime
you can avoid this
And, trying this last week with a
A J Stiles wrote:
But British phones do
not have a £ sign on the keypad; instead, we have a comment mark (#) key to
the right of the zero, usually called hash.
Just a note, the hash key in the US is called the pound key.
Doug
--
Ben Franklin quote:
Those who would give up Essential
On Sun, Jun 10, 2012 at 10:10:29PM +1200, Duncan Turnbull wrote:
Hi All
Just a quick check on the best way to ensure multiple cards/devices load in
the correct order.
Asterisk 1.8 with Sangoma A101 had no problems until we introduced an
Astribank.
root@pabx377:/etc/asterisk#
It seems like much of the problem with virtualizing Asterisk is getting it to
interface with DAHDI cards. Is that correct? As I'm not planning on using
such cards (I'm only interfacing with our broadband connection), would this
make virtualizing more feasible?
Richard
-Original
Hiers, Richard wrote:
It seems like much of the problem with virtualizing Asterisk is getting it to
interface with DAHDI cards. Is that correct? As I'm not planning on using
such cards (I'm only interfacing with our broadband connection), would this
make virtualizing more feasible?
I'd
We virtualize every asterisk install, and have achieved density levels of
80MB RAM per install of asterisk. We do it all day, every day.
As Chris wrote if you're putting it on shared hardware that you don't
control, just don't. If you control all of the hardware it's very doable.
Thanks
David
I would like to be able to use the dialing extension's voicemail box password
to authenticate or as a PIN code in the dialplan. Is there a best method for
doing this? I could use AGI scripting but I was hoping there was a built-in
dialplan means for
doing this. I have used VMAuthenticate but I
On Mon, Jun 11, 2012 at 8:34 AM, Chet W. Stevens
cwstev...@interact.ccsd.net wrote:
Also, related to this question; is there a best or recommended method to
determine the dialing extensions voice mail box? I have been using
variations of ${CUT(CHANNEL,-,1)} which does work but I just have to be
Hi All;
Any one used Digium IP Phones D40?
I need to know if they are stable with good voice quality? Comparing to Polycom
330, which is better? Let us talk frankly although I know that we have to
support Digium.
Regards
Bilal
--
On Mon, Jun 11, 2012 at 9:58 AM, bilal ghayyad bilmar...@yahoo.com wrote:
I need to know if they are stable with good voice quality? Comparing to
Polycom 330, which is better? Let us talk frankly although I know that we
have to support Digium.
Voice quality is great. I would choose the
Hello,
I would like to know the difference between encrypt the rtp and signaling
between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm
trying to understand whether an SBC could fit an Asterisk deployment
Thanks
--
On 12-06-11 12:58 PM, bilal ghayyad wrote:
Hi All;
Any one used Digium IP Phones D40?
I need to know if they are stable with good voice quality? Comparing to Polycom
330, which is better? Let us talk frankly although I know that we have to
support Digium.
I don't think these topics about
On 12-06-11 02:06 PM, Danny Dias wrote:
Hello,
I would like to know the difference between encrypt the rtp and signaling
between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm
trying to understand whether an SBC could fit an Asterisk deployment
What are you expecting the
On 06/11/2012 10:34 AM, Chet W. Stevens wrote:
I would like to be able to use the dialing extension's voicemail box
password to authenticate or as a PIN code in the dialplan. Is there a
best method for doing this? I could use AGI scripting but I was hoping
there was a built-in dialplan means for
On 12/06/2012, at 12:00 AM, Tzafrir Cohen wrote:
On Sun, Jun 10, 2012 at 10:10:29PM +1200, Duncan Turnbull wrote:
Hi All
Just a quick check on the best way to ensure multiple cards/devices load in
the correct order.
Asterisk 1.8 with Sangoma A101 had no problems until we introduced an
Dears;
I need to order Digium card and not able to know which one is the best quality?
Is it that of AEX with the end E or EF or P or B?
I saw those card that its slot is small (I think those that end by EF), are
they the best card?
Really I am caring to have a card that has echo cancelation
That's my question...the sbc provides security over trunking, right? The
same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of
add-value to an Asterisk deployment?
El 11/06/2012 20:20, Paul Belanger paul.belan...@polybeacon.com
escribió:
On 12-06-11 02:06 PM, Danny Dias
On 06/11/2012 01:48 PM, bilal ghayyad wrote:
I need to order Digium card and not able to know which one is the best quality?
Is it that of AEX with the end E or EF or P or B?
Digium does not produce cards of differing quality levels; we strive to
have all of our cards be produced with the
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 defirenciate from internal call or external call?
Thanks,
motty
--
Hi
As far as I know Alert-Info is as far as vendor specific extension
to SIP used by CISCO VOIP-gateways only. Didn't noticed any other
vendors to support that. Software clients neither. So such trick is
only usable in conjunction with CISCO.
Anyway, wait another answer, probably somebody knows
On Wed, 6 Jun 2012 16:57:11 -0400
Hai Nguyen h...@jazinga.com wrote:
A calls B. B attended-transfers the call to C using (polycom, cisco)
phone's transfer button. C does not answer the call. A gets B's
voicemail. However, if B blind-transferred the call to C and C did
not answer the call, A
On Mon, Jun 11, 2012 at 6:12 PM, motty.cruz motty.c...@gmail.com wrote:
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
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