at the moment we have a ToIP trunk for incoming and outgoing calls, using OpenSIPS as SIP router for registration and calls routing. It works well but it lacks some functionality like transcoding and call-pickup (“Replaces:”) support.
I’m wondering if i can add an Asterisk box between my OpenSIPS and the provider eSBC (AudioCodes Median) to support Replaces and call transfers.
Meanwhile, i’m happy to get suggestions or tips related to this need.
for transcoding use GitHub - sipwise/rtpengine: The Sipwise media proxy for Kamailio it has better scaling than Asterisk and support more codec
and what do you mean that OpenSIPS do not support Replaces
we uses Kamailio for Pickup in our setup and that is almost identical with OpenSIPS
or is it because you ToIP do not allow Replaces
depending on you need for scalability you might also want to look at SIMS
but yes you can use an asterisk to perform Pickup, Transfer and transcoding
Thanks for your kind and interesting reply!
Yes, as you said, our SBC doesn’t support Replaces, so i need to implement something between to let my users doing call pickups and transfers.
I’ll take a look to SIMS, looks interesting.
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