07.03.2015 1:21, Kevin Harwell пишет:
On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
07.03.2015 0:24, Kevin Harwell пишет:
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov
serov@gmail.com mailto:serov@gmail.com wrote:
Ira wrote:
Hello John,
Friday, March 6, 2015, 12:34:42 PM, you wrote:
Find a HPT5720 with expansion chassis on eBay for under $50,
load AstLinux ( instructions at AstLinux.org ) Move your
Digium card and your confs , fix up any differences from your
But given that means buying an old
John
I will have to get one of these and give this a try. Thanks for sharing.
Thanks
Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.)
616-855-1030 Ext. 2003
From: John Novack SCII jnov...@stromberg-carlson.org
Sent: Friday, March
Iran
For the kind of loads and low cost you are talking with 2 FXO, 2FXS and
SIP the Grandstream UMC6102 is low power feature rich and easy to maintain.
Check it out -
http://www.grandstream.com/index.php/products/ip-voice-telephony/ip-pbx-solu
tions/ucm61xx
If you do choose to use the
BTW, the allow=!all is equivalent to disallow=all, so you can drop the
disallow line.
On Thu, Mar 5, 2015 at 7:26 PM, Sonny Rajagopalan
sonny.rajagopa...@gmail.com wrote:
OK. I think I found the issue.
The key is to add
rtp_symmetric=yes
Here's what my final configuration looks like:
Kris Stark wrote:
Unfortunately, I was never asked about this to enough detail to be
able to tell them how to set up the music, and as a result I have an
eight minute file with several different messages all tied together
into that one file.
You can use Audacity to break that file into
I am dealing with a FreePBX generated dialplan. I have been following
the processing traces attempting to make use of the advice I received
here respecting setting a custom ring tone. I have discovered that
the context I am using for incoming calls is not used at all during a
blind transfer.
Hi
I plan to host Asterisk instances on AWS/EC2 servers.
Requirement is to run asterisk instance with transcoding (g.729 + g.711) and
full recording. Number of concurrent calls expected are 500+. 2 instances will
be configured for 100% redundancy. Heart beat will be used to determine active
Why use Amazon? With that kind of load I would want dedicated servers.
Call Rackspace or Softlayer.
j
On 03/06/2015 11:59 AM, Amit Patkar wrote:
Hi
I plan to host Asterisk instances on AWS/EC2 servers.
Requirement is to run asterisk instance with transcoding (g.729 +
g.711) and full
*friends help me *
*cant get incoming calls in asterisk*
*(when i connect **80081 in softphone ---every thing is ok**)*
*--- SIP read from UDP:200.152.125.221:5060 http://200.152.125.221:5060
---*
*INVITE sip:80081@10.47.10.10:5060 http://sip:80081@10.47.10.10:5060
SIP/2.0*
*Record-Route:
Hello Asterisk,
Back in 2009 I built a small Intel Atom based computer running
Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs
line and six or so SIP numbers. So basically no load. I'm
feeling like it's time to build another machine. It's probably
silly, but it's been six
Hello.
Asterisk 13.2.
I transfer configs from chan_sip to res_pjsip.
In chan_sip i have match_auth_username=yes and have nothing in pjsip.
I have a lot of endpoints and registrations on same SIP server. And it's
problem in pjsip now. Is not it?
I requesting to add new value for endpoint
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com wrote:
Hello.
Asterisk 13.2.
I transfer configs from chan_sip to res_pjsip.
In chan_sip i have match_auth_username=yes and have nothing in pjsip.
I have a lot of endpoints and registrations on same SIP server. And it's
07.03.2015 0:24, Kevin Harwell пишет:
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
Hello.
Asterisk 13.2.
I transfer configs from chan_sip to res_pjsip.
In chan_sip i have match_auth_username=yes and have nothing in
Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your
older version of Asterisk to the fairly current version 11 currently available with AstLinux.
Use the GUI to edit
If you are really wanting to build something on Raspberry Pi or similar ARM
platform, you could also take a look at Elastix for ARM.
http://www.elastix.com/en/downloads/ Elastix is a fully integrated
platform, and includes the majority of necessary components in one
installation.
The new
Hello.
I continue to transfer chan_sip to pjsip.
Friend in chan_sip can has options:
deny=0.0.0.0/0.0.0.0
permit=192.168.0.1
pjsip offer to use global ACL without relation to any andpoint.
My task is restriction via IP to registering in certain endpoint.
Different rules to different
Hello John,
Friday, March 6, 2015, 12:34:42 PM, you wrote:
Find a HPT5720 with expansion chassis on eBay for under $50,
load AstLinux ( instructions at AstLinux.org ) Move your
Digium card and your confs , fix up any differences from your
But given that means buying an old computer, why
On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com wrote:
07.03.2015 0:24, Kevin Harwell пишет:
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com
wrote:
Hello.
Asterisk 13.2.
I transfer configs from chan_sip to res_pjsip.
In chan_sip i have
Hi Jeff
Are you aware of any challenges of hosting it on AWS? It will help me to
work out alternate plan. Is there any recommendation? Should I split it
to multiple instances and balance traffic across multiple small server
instances? I can use Kamailio to balance traffic.
I see many posts
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