Hi Friends,
I encountered a strange issue.
I am running Asterisk 11.8.1 on Cent OS with no firewall running.
It has 3 NIC interfaces.
in my sip.conf I have
allowguest=yes
bindaddr=0.0.0.0
udpbindaddr = 0.0.0.0
But my Asterisk instance does not pick the call at all.
When I check the listening
Hello,
Pardon, this might be off-topic. I'm reading:
http://en.wikipedia.org/wiki/Network_switch
For a setup of ~5 agents, would I be wrong in thinking that a generic 16
port unmanaged switch would fit the bill?
The first model to come up for me in an Amazon search is:
On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote:
For a very basic setup it would work, but I would suggest POE at a
minimum, and vlan support if possible.
thanks for the recomendations :)
-Thufir
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Nick Awesome wrote:
Hay guys, have question.
When I do regular dial I use
$this-AGI-get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
to get all contacts of current endpoint and so I dial to all phones
at once,
but if I exec QUEUE, I have just one phone rings, seems like
Works, thank you!
On Feb 23, 2015, at 7:11 PM, Joshua Colp jc...@digium.com wrote:
Nick Awesome wrote:
Hay guys, have question.
When I do regular dial I use
$this-AGI-get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
to get all contacts of current endpoint and so I
Starting with Asterisk 13.1 we are seeing this WARNING
messages a lot in our logs and console:
WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type
frames with SIP write)
We found that line in function sip_write inside chan_sip.c.
In our previous
Hay guys, have question.
When I do regular dial I use
$this-AGI-get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
to get all contacts of current endpoint and so I dial to all phones at once,
but if I exec QUEUE, I have just one phone rings, seems like it take first one
as
thank you, we are using the same configuration files in 13, same setup, just
different asterisk version. we just dont see the msgs in the console/logs, it
is the same exact voice traffic on both asterisk versions
is that something that you set on/off? if that is the case how can it be done?
Hello everyone,
I am trying to activate Music On Hold using DB on Asterisk 13.
It works fine but in order to use new Music On hold definitions I have to
reload the moh module.
- The following is my configuration in extconfig.conf - I added the
following line:
musiconhold.conf =
Hay guys, got trouble with registration with cisco 7975
Here is the debug :
--- Received SIP request (576 bytes) from UDP:192.168.1.61:49533 ---
REGISTER sip:192.168.1.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381
From:
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