Hello,
With asterisk 12 improvements, is it now possible to bind an asterisk SIP
stack to several ports ?
For instance, to both emit or listen on ports 5060 and 5062.
If positive, any hint on how to get more detail about this ?
Regards
--
Olivier wrote:
Hello,
With asterisk 12 improvements, is it now possible to bind an asterisk
SIP stack to several ports ?
For instance, to both emit or listen on ports 5060 and 5062.
chan_sip does not support this but chan_pjsip does by configuring
multiple transports, each using a different
On 01/08/14 10:56, Olli Heiskanen wrote:
Hi,
I got ahead with my setup, this post helped me much:
http://forums.digium.com/viewtopic.php?f=1t=90167sid=66fdf8cc4be5d955ba584e989a23442f
At least the avpf setting had to be removed from sip.conf and put in
the realtime db table, defined per
Great !
I'm gonna it try ASAP !
Is there another way (ie not using different ports) to get several trunks
to a given ITSP ?
Let me explain this a bit further.
My setup is:
ITSP SIP Asterisk Phones
For various reasons, I want my Asterisk box to have several trunks/SIP
account with
Olivier wrote:
Great !
I'm gonna it try ASAP !
Is there another way (ie not using different ports) to get several
trunks to a given ITSP ?
Let me explain this a bit further.
My setup is:
ITSP SIP Asterisk Phones
For various reasons, I want my Asterisk box to have several
Hi all,
I've got a few devices, SPA112's and SPA8000's, that are giving me problems.
Each device has a separate SIP credential for each port, but sometimes, only a
few of the ports register.
So, the device will be running fine for a while, then suddenly one or more of
the ports will become
I've got a few devices, SPA112's and SPA8000's, that are giving me
problems.
Each device has a separate SIP credential for each port, but
sometimes, only a
few of the ports register.
So, the device will be running fine for a while, then suddenly one or
more of
the ports will become
On Tuesday, August 05, 2014 11:01:01 AM Kevin Larsen wrote:
I've got a few devices, SPA112's and SPA8000's, that are giving me problems.
Each device has a separate SIP credential for each port, but
sometimes, only a
few of the ports register.
So, the device will be running fine for a
On 5 Aug 2014, at 17:10, Mike Diehl mdiehlena...@gmail.com wrote:
All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10.
If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly…
Freezing and requiring power-cycle, clocks stopping (and showing minus
On Wed, Jul 30, 2014 at 10:29:19PM +0200, Anthony Azzopardi wrote:
Hello asterisk-users,
I need to compile dahdi and then export it to another system. I managed to
do this with DESTDIR=/root/destDir, then make a tar file and extract in / of
the other system. However the module is not
On Tuesday, August 05, 2014 05:19:55 PM Steven Howes wrote:
On 5 Aug 2014, at 17:10, Mike Diehl mdiehlena...@gmail.com wrote:
All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10.
If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly…
Freezing
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.10.0-rc1
DAHDI-Tools-v2.10.0-rc1
dahdi-linux-complete-2.10.0-rc1+2.10.0-rc1
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
Working on a paging system for one of my sites and running into something
I can't believe is this hard. In one of the zones, they want to have three
different extensions ring over the pa system, using it as a loud ringer.
Now the paging system does have a loud ringer built in and I can easily
Hello. Help me please.
Tell me how to implement this action:
How to play a sound file or perform any action the user to an incoming
call after a conditional transfer?
Example. Subscriber A calls subscriber B. Subscriber B transfers the
call to the subscriber C (previously talking to him).
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