Hi Ghaith,
Force_send_socket() does not accept a variable,but you can use instead
the $fs (force socket) var which does accept variables :
$var(a) = "x.x.x.x:xx";
$fs = $var(a) ;
Regards,
Bogdan
Ghaith ALKAYYEM wrote:
> Hi,
>
> Is it possible also to make bridging dependent on a variable value by
> passing a variable as a parameter to force_send_socket() as following:
>
> $var(a) = "x.x.x.x:xx";
> force_send_socket("$var(a)");
>
> because the above configuration gave me an error but when I used the
> variable in xlog function it was okay:
> xlog("$var(a)");
>
> I might do some code modification in this regard.
>
> Regards.
>
> On Mon, 2009-08-24 at 18:03 +0300, Bogdan-Andrei Iancu wrote:
>
>> Hi Matthew,
>>
>> There 2 things when comes bridging:
>>
>> 1) signalling part - selecting the proper outbound interface (private or
>> public)
>> a) this can be automatically done by opensips (based on the
>> destination IP) if you enable the mhomed parameter in core ; this is
>> simple by not so efficient
>>
>> b) you can do it manually, by selecting from script the correct
>> interface - see the force_send_socket() function
>>
>> 2) media part
>> a) rtpproxy - when enabling RTPproxy (at request and reply time)
>> you can explicitly select which interface to use (see the e and i flags
>> - http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id271362)
>>
>>
>> Best regards,
>> Bogdan
>>
>> Matthew S. Crocker wrote:
>>
>>> Hello,
>>>
>>> I'm brand new to OpenSIPS, just going through the make process now.
>>>
>>> I need to configure OpenSIPS to act like a SBC for some SIP trunks coming
>>> off a VoIP switch. Where should I look for Documentation/Examples of a
>>> working config?
>>>
>>> Here is my scenario:
>>>
>>> OpenSIPS has two interfaces, private & public.
>>> VoIP Gateway is on private LAN with no gateway configured (it can only talk
>>> to local machines, no routing)
>>>
>>> End user has an Asterisk server on a private lan behind their firewall (NAT)
>>>
>>> I need to configure OpenSIPS to listen for SIP messages on :5060 from the
>>> end user firewall. It then need to rewrite the SIP message and send it to
>>> the Gateway. The Gateway would see the messages coming from the internal
>>> IP of the OpenSIPS server. Once all of the SIP messages get processed I
>>> then need the OpenSIPS server to proxy the RTP streams (plan on using
>>> mediaproxy) between the Asterisk server and VoIP Gateway.
>>>
>>> Any helpful hints on where to look?
>>>
>>> -Matt
>>>
>>>
>>>
>>>
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>>
>
>
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