Do you want from your classic skype client or from former lync (now
skype for business) -- just to avoid confusions given the close naming
they have these days?
Integration with Lync should work given you use tcp or tls.
To route to a Jabber server you will have to use Asterisk or FreeSwitch
as
I want to be able to route sip invites from my on prem skype to either an
on prem jabber server or out to office 365. It depends on who the recipient
is. I am sure Kamailio can do this but I am absolutely at a loss on where
to start.
I have tried a few different configs but I basically have to
Folks,
I'm working on a 5 node network in the following topology:
Host1 - Edge Proxy 1 --- Core Proxy - Edge Proxy 2 - Host2
The proxy machines contain SIP Server installations, the hosts contain user
agent software. I am able to make a call end to end between the hosts, but when
I look at
On 11/19/10 8:27 PM, Iñaki Baz Castillo wrote:
2010/11/17 Daniel-Constantin Mierlamico...@gmail.com:
I made an easy-to-do tutorial where all the SIP routing logic is implemented
in a Lua script (including authentication, accounting, registrar, user
location). You can read it at:
Daniel-Constantin Mierla wrote:
On 11/19/10 8:27 PM, Iñaki Baz Castillo wrote:
2010/11/17 Daniel-Constantin Mierlamico...@gmail.com:
I made an easy-to-do tutorial where all the SIP routing logic is
implemented
in a Lua script (including authentication, accounting, registrar, user
On 11/20/10 10:24 AM, Jeremya wrote:
Daniel-Constantin Mierla wrote:
On 11/19/10 8:27 PM, Iñaki Baz Castillo wrote:
2010/11/17 Daniel-Constantin Mierlamico...@gmail.com:
I made an easy-to-do tutorial where all the SIP routing logic is
implemented
in a Lua script (including authentication,
On 11/20/10 10:27 AM, Jon Bonilla (Manwe) wrote:
El Sat, 20 Nov 2010 10:15:27 +0100
Daniel-Constantin Mierlamico...@gmail.com escribió:
This should not be that complex, there is no function name that has '.'
in order to have current conflicts to solve. My concern is related to
modules with
2010/11/17 Daniel-Constantin Mierla mico...@gmail.com:
I made an easy-to-do tutorial where all the SIP routing logic is implemented
in a Lua script (including authentication, accounting, registrar, user
location). You can read it at:
http://asipto.com/u/h
sr.tm.t_check_trans()
Hello,
during last days I spent some time to extend the native API exported to
Lua. Many more functions exported by core and modules can be called from
embedded Lua scripts. Note that you get also access to psedo-variable
operations and you can call all functions exported by modules that have
Definitely cool!
Thanks,
Carsten
2010/11/17 Daniel-Constantin Mierla mico...@gmail.com:
Hello,
during last days I spent some time to extend the native API exported to Lua.
Many more functions exported by core and modules can be called from embedded
Lua scripts. Note that you get also access
I think you have multiple issues:
(not sure if I am right as I have to guess your network setup)
1. If Kamailio is an Application Layer Gateway between the public and
the internal network, then of course Asterisk should listen only on the
internal interface. Thus in sip.conf add:
Hello All,
I'm using Kamailio as a Border controller for my VoIP Research project at my
school.
The problem I'm facing is Kamailio routes the traffic to the private network
where my asterisk machine is listening.
The asterisk machine responds to the Kamailio using the public network but
not the
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