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[Antony.Stone at freesw...]

Hi.

I've recently started using FreeSwitch, after being familiar with Asterisk for
many years.

I'm wondering whether FreeSwitch has an equivalent to Asterisk's facility for
dialling out to a SIP server using credentials supplied as part of the dial
command, instead of needing a pre-defined gateway?

In an Asterisk dialplan, you can say something like

Dial(SIP/${Destination}:${SIPpass}::${SIPuser}@${SIPhost})

and that will dial to the Destination number on SIPhost using the SIP
credentials SIPuser/SIPpass. Nothing needs to be configured in advance.

Can FreeSwitch do something like that too?


Thanks,


Antony.

--
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[kaduww at gmail.com]

Hey,

It can, set the variables sip_auth_username and sip_auth_password


bridge {sip_auth_username=<your_user_name>,sip_auth_password=<your_password>}sofia/external/${destination_number}@12.34.56.78 ([email]destination_number%7D@12.34.56.78[/email])

Here you have the list of variables you can use https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List


Em qui., 14 de out. de 2021 às 21:32, Antony Stone <Antony.Stone@freeswitch.open.source.it (Antony.Stone@freeswitch.open.source.it)> escreveu: