[asterisk-users] asterisk-users Digest, Vol 177, Issue 11

Joshua C. Colp jcolp at digium.com
Tue May 28 08:05:20 CDT 2019

On Sat, May 25, 2019, at 2:34 PM, Saint Michael wrote:
> Joshua
> Is there a way in PJSIP to send the audio between the parties always, 
> unless one of the parties is behind a NAT?
> A session refresh would work.
> That my only problem with PJSIP. This is routine in the old sip channel.

Any such functionality would be documented on the wiki[1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip

Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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