SIP server configuration for SDP


(Mark Xavier) #1

Is it possible for the sip server to reply with 180 without SDP?
I am currently using raspbx.
I want a pjsip extension to ring in without a ringer tone.


(Joshua C. Colp) #2

PJSIP support in Asterisk already responds without SDP for 180, it’s then up to the caller to produce the ringing.


(Mark Xavier) #3

The sip server is replying to the pjsip client’s call with 183+SDP and then sends it RTP
Is this a configuration in freepbx that I can alter?


(Joshua C. Colp) #4

I can’t speak for FreePBX as to what it is doing. You’d also need to describe the actual call scenario.


(David55) #5

I think the dialplan would have to call Progress() for that to happen, so don’t do that.

Note that if the B side sends 183, you really should be using early media on the A side.


(Mark Xavier) #6

is that possible through freepbx administration page?


#7

I’m very confused by your comments. Are you trying to change the behavior of internal calls, outgoing calls, or incoming calls? Please give specific examples. Are you trying to change what the caller hears, what the called party hears, or are you trying to change signaling or RTP for some technical reason e.g. to affect the behavior of an external automated system?


(Mark Xavier) #8

I am trying to change the behavior of incoming calls. I hope is that the caller does not hear a ringtone.
In my scenario, a sensor initiates a call to a ring group ( of sip clients) using a pjsip client (from fanvil).


#9

Send you incoming call to a custom context that has a priority 1 of Answer(), the rest of that context is up to you


(David55) #10

That pretty much guarantees they will get media (ordinary rather the early) even though it may be silent media.


#11

Set your Ring Group to Play Music On Hold with the music being silent.


(Mark Xavier) #13

thanks that works