[asterisk-users] SIP Strict Routing and canreinvite

[Mindaugas Kezys]

Hello,

 

I want to send Media outside Asterisk server, e.g. between peers.

 

In CLI I see:

 

.  [Jun  8 13:13:58] VERBOSE[19112] logger.c:     -- Native bridging
SIP/5060-b7dc5218 and SIP/prov12-09ad3888       

.  [Jun  8 13:13:58] DEBUG[19112] chan_sip.c: Strict routing enforced for
session 3ad367ee48778d2c523a60e62ae86822 at 85.113.41.129   

 

And media still goes through Asterisk.

 

Why is that?

 

Why strict routing is enforced?

 

 

Regards,

Mindaugas Kezys

http://www.kolmisoft.com

VoIP Billing and Routing Solutions

 

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