Unable to turn on EarlyRTP

[BrunoVicious]

I am unable to turn on EarlyRTP even when it's configured in the sccp.config.

SEP64168D5028C1: (getEmptyChannel) on line:101, maybe_c:
-- SEP64168D5028C1: Getting the active channel on device.
-- SEP64168D5028C1: No active channel on device.
-- SEP64168D5028C1: Set the active channel 1 on device
-- SCCP/101-00000001: (setEarlyRTP) OFF
-- 13: New channel number: 1 on line 101
-- SCCP: (pbx_channel_allocate) try to allocate Outbound channel on line: 101
-- SCCP: cid_num: 101
-- SCCP: cid_name: Line 101
-- SCCP: accountcode: 101
-- SCCP: exten:
-- SCCP: context: from-internal
-- SCCP: amaflags: 0
-- SCCP: chan/call: SCCP/101-00000001
-- SCCP: combined audio caps: [slin16, ulaw/64k, alaw/64k, ilbc, g729b, g729ab, g729, g729a]
-- SCCP: reduced audio prefs: [ulaw/64k]
-- SCCP: combined video caps: []
-- SCCP: reduced video prefs: []

chan-sccp version: How do I find this?

Current Asterisk Version: 16.15.1

Expected behavior:

I am trying to get a ringback tone when I call out on the trunk line. At the moment it only work when I call internal users but not when I use the SIP trunk line.

[dkgroot]

@BrunoVicious

• In the log above, the message "-- SCCP/101-00000001: (setEarlyRTP) OFF" is a little disconcerting, and i am wondering where this is coming from.
• Getting earlyrtp to work on outbound calls has proven to be a little tricky in the past, it requires carefull cooperation between all channels on the path (phone, sccp, pbx, sip/pjsip, provider, endpoint)
• Are you using chan-pjsip or chan-sip (legacy) ? Both channel drivers need to cooperate carefully, to make earlyrtp happen. I have mostly worked with chan-sip and gotten it to work on outbound calls. I never really got around to making the jump to pjsip, to be honest. But that is irrelevant, it ought to work with any channel driver, so it has to be fixed.
• We need to figure out if earlyrtp / earlymedia is active at the start of the call (sccp indicate), if there is any rtp traffic flowing (tcpdump/wireshark), if the codecs choosing on the path are compatible (asterisk loggin). So we have a bit of work to do.
• Let's start with that notice above. Can you provide the call log for one call with:
  • core set verbose 5
  • core set deug 1
  • sccp debug core, channel, pbx, rtp, codec, indicate