I am working to set up a group of multicastRTP auto page messages that are executed from the dial plan using call files.
My live paging works great. 1 simple line in extensions.conf:
exten => 50,1,Dial(MulticastRTP/basic/126.96.36.199:32050)
I have the everything set to create the call file with the extension to set up the multicast Channel and then connect the extension to playback the audio. It functions but the audio is so choppy it is non functional.
I suspect is is a transcoding issue because when I “core show channel multicastRTP/xxxxxx” it is different if it is a live page vs a call file set up page.
On a live page the phone initiates the call with pjsip setting the codec to ulaw, the phone is listening on ulaw so the multicastRTP channel is set up as ULAW and all works fine.
On the call file version the multicast is initiated with the c(ulaw) option but the channel is still set up as slin requiring the codec transcodeing.
I have gone so far as to try to modify the chan_rpt code to set the default codec to ulaw but no luck.
Anyone had and solved this issue?