Thanks SamyGo and Mitul for your prompt responses.
I have been vested with the responsibility to evaluate Asterisk for a VOIP
solution. I was just going through couple of documents and got impressed by
the features it has to offer. Our user base is around 15000 and the system
should support 1000
15k users are quite a big number. To my clients with a large user base I
advice always to partition the load on multiple servers. This has a list of
advantages, like the ability to power cycle a node without impacting all
your users, easier debug and tests of problems and solutions, abiity to
15k users is not a big deal, however 1000 concurrent calls surely is.
Asterisk will start blurring up on 250 concurrency.
In short whatever you are needing to setup needs quite a few of other
components like OpenSIPs or FreeSWITCH etc.
If its VoIP (SIP) Only then you might be better off using
Thanks Leandro for your reply. See my comments inline.
On Mon, Jul 23, 2012 at 12:57 PM, Leandro Dardini ldard...@gmail.comwrote:
15k users are quite a big number. To my clients with a large user base I
advice always to partition the load on multiple servers. This has a list of
advantages,
Thanks Mitul. But according to some performance data available on the
Internet, 15000 users is a big deal it seems, if Asterisk has to function
as a SIP registrar. Sure, 1000 concurrent calls is a big deal, and we are
planning to balance the load across three server hardware -- HP DL 380s.
No, it will not have and if You just put and.ulaw. You should
correct file say.conf - there are rules how to read numbers, and You
should add and there if You want to hear it.
2012/7/23 נפתלי מאיר nafma...@gmail.com:
It`s not will to be: ; one - thousand - two - hundred - and - thirty - four
??
On Mon, Jul 23, 2012 at 10:55:54AM +0300, נפתלי מאיר wrote:
Hello,
I use the SayNumber() with variable.
for example the number 1234 - asterisk play the number without and.
-- Executing [888@from-internal:1] Set(SIP/103-035d,
LANGUAGE=en) in new stack
-- Executing
Thank you guys.
I found say.conf example file with and.
Naftali
On Mon, Jul 23, 2012 at 1:38 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Jul 23, 2012 at 10:55:54AM +0300, נפתלי מאיר wrote:
Hello,
I use the SayNumber() with variable.
for example the number 1234 -
Hi everyone,
we're trying to setup a fax gateway from PRI to SIP. Our test setup
consists of two asterisk 10 machines connected over PRI and a third one
which is acting as a fax endpoint over SIP.
On the machine which is receiving over PRI, we see the following message:
[Jul 23 12:54:23]
www.checkbox.cc
On Friday, March 16, 2012, white hat wrote:
I had many of the same problems with sip station. If you just need sip
termination, Check out flow route. The service just seems to work properly
for me, and they respond to tickets. You can open up new cases through
their site.
Answers in text.
2012/7/23 Kannan vasdevelo...@gmail.com
Thanks Leandro for your reply. See my comments inline.
On Mon, Jul 23, 2012 at 12:57 PM, Leandro Dardini ldard...@gmail.comwrote:
15k users are quite a big number. To my clients with a large user base I
advice always to partition
On 07/23/2012 06:30 AM, Paul Goldbaum wrote:
Hi everyone,
we're trying to setup a fax gateway from PRI to SIP. Our test setup
consists of two asterisk 10 machines connected over PRI and a third one
which is acting as a fax endpoint over SIP.
This 'third' system is the one responsible for
Just a quick heads-up for anyone thinking of trying out the new Digium Phones:
If you are not using DPMA (and we can't, because it is only made available in
binary form and our software procurement policy forbids this. I'll hold out
for an Open alternative) then you can still use these
For private use the RPI is really cool for that, though I am not yet sure
if it works 100% without problems - at least it did in my latest tests.
Anyone has any hint on the call quality or if Asterisk does any kind of
transcoding of the audio?
2012/7/21 Mike ispbuil...@gmail.com
On 12-07-21
On 07/23/2012 10:56 AM, Stefan at WPF wrote:
For private use the RPI is really cool for that, though I am not yet
sure if it works 100% without problems - at least it did in my latest tests.
Anyone has any hint on the call quality or if Asterisk does any kind of
transcoding of the audio?
If
Hello,
I tried Asterisk Confbridge with raspberry pi without audio issue.
Asterisk was compiled from sources.
http://www.voztovoice.org/?q=node/553
Regards
--
_
-- Bandwidth and Colocation Provided by
Hello,
I'm trying get fax working over VOIP lines. I'm running Asterisk 1.8.12
server, working fine, however I would like to get rid of our anolog fax
lines and integrate with our fax to our Asterisk Server.
Any recomendataion from this list? I had done some research but nothing
solid.
Hey mailinglist,
my problem still exists and I need a little bit help.
When I start Asterisk, I do the following:
asterisk -rv
originate SIP/IMSI123456789101112 application MusicOnHold
Perhaps this will help you:
*CLI sip show peers
Name/username Host
On 07/23/2012 09:23 AM, motty.cruz wrote:
Hello,
I'm trying get fax working over VOIP lines. I'm running Asterisk 1.8.12
server, working fine, however I would like to get rid of our anolog fax
lines and integrate with our fax to our Asterisk Server.
Any recomendataion from this list? I had done
Hi Stelios,
Thanks for the response.
I take the following excerpt from your response. --- You can, but usually
for virtual/hosted pbx's you need an additional
layer of management software or a lot of copy paste
Could you please elaborate on that? Do need to modify Asterisk or there
exists some
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