[SR-Users] About routing calls

[Daniel-Constantin Mierla]

Hello,

your request is not clear: do you want Kamailio not to send the SIP
traffic to Asterisk, but Asterisk still to get the RTP (audio) stream of
the call?

Cheers,
Daniel

On 28.06.20 23:41, ahmed moghazy wrote:
> Excuse me i already made integration for kamailio with asterisk but  i want to make sip calls through kamailio and media fuctions through asterisk , i want kamailio to make calls away from asterisk , i just wanted asterisk to use its dahdi module .’
>  Thank you.
>
-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla