Hi all.
I've installaed a PIAF with Asterisk 13.0 on a ESX virtual machine on a DELL
node.
As usual I loaded g729 codec in modules folder but I notice very high
translation time.
Over several PIAF with Asterisk 1.8.X (always in a virtual environment) I
always had for example 601 between G729
Show him this freaking thread, or else ask him to prove it otherwise.
We all here have decades of exp dealing with asterisk.
Mitul
On 07-Apr-2015 7:27 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com wrote:
Dear Mitul,
I already told my boss about it, I really want a single box, no virtual,
but
Dear all,
Is anyone has experience making Asterisk server with virtual server OPEN-VZ
(in proxmox 3.4 box) ?
My boss want to build a production server with it, and it will have +/- 300
sip user (concurrent call maybe 150 call)
Is it good to go, or not ?
I really hope someone who have
Why not use just one single box and create 300 sip clients having 150 odd
con calls. OpenVZ might not be a good idea for this sort of volume.
Mitul
On 07-Apr-2015 7:12 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com wrote:
Dear all,
Is anyone has experience making Asterisk server with virtual
Dear all,
Is anyone has experience making Asterisk server with virtual server OPEN-VZ
(in proxmox 3.4 box) ?
My boss want to build a production server with it, and it will have +/- 300
sip user (concurrent call maybe 150 call)
Is it good to go, or not ?
I really hope someone who have
Dear Mitul,
I already told my boss about it, I really want a single box, no virtual,
but my boss insist.
He said that openvz use less resource then KVM (or other virtual for cloud).
I really need a solid analysis to argue with him.
On the other hand, dahdi cannot be installed in openvz virtual
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On 04/07/2015 09:41 PM, Ikka Tirtawidjaja wrote:
Dear all,
Is anyone has experience making Asterisk server with virtual
server OPEN-VZ (in proxmox 3.4 box) ?
My boss want to build a production server with it, and it will have
+/- 300 sip user
I have several large customers (200+ extensions) running on vSphere without
issue. Not sure about OpenVZ, thought.
2015-04-07 11:36 GMT-03:00 Mitul Limbani mi...@enterux.in:
Show him this freaking thread, or else ask him to prove it otherwise.
We all here have decades of exp dealing with
PBX =! CC my friend.
150 Conc Calls for CC agent is going to be far more expensive then running
200 extn PBX doing hardly 20 Conc Calls.
Load is way too diff.
On 07-Apr-2015 8:18 PM, Vinicius Fontes vinic...@aittelecom.com.br
wrote:
I have several large customers (200+ extensions) running on
I guess best way for your boss to learn is to deploy a box once and get
bombed and then follow what ppl said here.
Both modes u should be the happy guy u see, u will get paid twice for same
work !!!
Mitul
On 07-Apr-2015 7:27 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com wrote:
Dear Mitul,
I
Dear Guenther B.
This server is not for call center. Its for office and appartment with +/-
900 sip users.
The asterisk server will be split to 3 OpenVZ Virtual server in 1 proxmox
server (but they will have clustering server in another proxmox server) and
1 database server (mysql), and it also
On 04/07/2015 10:48 AM, Johan Wilfer wrote:
Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev:
Dear all,
Is anyone has experience making Asterisk server with virtual server
OPEN-VZ (in proxmox 3.4 box) ?
My boss want to build a production server with it, and it will have +/-
300 sip user
Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev:
Dear all,
Is anyone has experience making Asterisk server with virtual server
OPEN-VZ (in proxmox 3.4 box) ?
My boss want to build a production server with it, and it will have +/-
300 sip user (concurrent call maybe 150 call)
As long as you
With that kind of load, your users shall start complaining about choppy
audio or voice clarity on random occasions, and you wont have a clue where
to look for the problem.
Regards,
Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W),
Hi all,
We are running a fax2email service based on asterisk 1.8.18.0, and we are
currently trying out asterisk 1.8.32.2 in our labs. We get the following
error when sending faxes out:
[Apr 7 14:34:20] ERROR[16653]: res_fax.c:2121 sendfax_exec: 'modems'
setting 'V17,V27,V29' is incompatible with
Den 2015-04-07 20:47, Mitul Limbani skrev:
With that kind of load, your users shall start complaining about choppy
audio or voice clarity on random occasions, and you wont have a clue
where to look for the problem.
That's another issue thought and is not different on a dedicated server.
With
Does anyone know anything about the Fidelio and Mitel protocol for hotel /
motel?
Are these industry standards or proprietary formats?
Are there open standards for communication with Hotel management
software's that could be used in conjunction with a custom asterisk
deployment?
I am trying to collect enough information about an problem a client is having
with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20,
and we upgraded to 11.15.0 and then to 11.17.0 with no solution.
Background: this client is a telemarketing call-center that generates
El 07/04/15 a las 17:38, Alex Villacís Lasso escribió:
I am trying to collect enough information about an problem a client is having
with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20,
and we upgraded to 11.15.0 and then to 11.17.0 with no solution.
Background: this
Hi Dmitriy and others and thanks for your help so far.
The option match_auth_username=yes seems to have had no effect. From my
reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying
Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
it does work. For prosperity, the SIP service is through Internode.
Here is my extensions.conf file:
exten = s,1,Set(thedid=${SIP_HEADER(TO)}); ignore this one
exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
exten =
Hi, Andrew.
You are trying to solve two tasks: definition through what line the call
came and a beautiful display of this information.
1. definition through what line the call came. If the username and
password for inbound and outbound registration the same, then try the
following:
a) delete
On 4/7/15 7:48 PM, Andrew Galdes wrote:
Hi Dmitriy and others and thanks for your help so far.
The option match_auth_username=yes seems to have had no effect. From
my reading, this option will try to match the username of the incoming
SIP account to a section heading. If that is how it must
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