I’m trying to use AudioSockets in Asterisk 18.9.0, but Asterisk doesn’t seem to understand my AudioSocket endpoints.
-- Executing [518@all-utilities:2] Dial("PJSIP/HostedTestTrunk-00000015", "AudioSocket/192.168.1.151:62021/1a332889-81bf-4903-be69-64cd54f51ca2") in new stack
[Aug 31 11:31:59] WARNING[253324][C-00000016]: channel.c:6244 request_channel: No channel type registered for 'AudioSocket'
[Aug 31 11:31:59] WARNING[253324][C-00000016]: app_dial.c:2600 dial_exec_full: Unable to create channel of type 'AudioSocket' (cause 66 - Channel not implemented)
-- No devices or endpoints to dial (technology/resource)
AudioSocket doesn’t appear in my channel types:
HostedTest1*CLI> core show channeltypes
Type Description Devicestate Presencestate Indications Transfer
------------- ------------- ------------- ------------- ------------- -------------
Recorder Bridge Media Recording Channel Driver no no yes no
Announcer Bridge Media Announcing Channel Driver no no yes no
CBAnn Conference Bridge Announcing Channel no no yes no
CBRec Conference Bridge Recording Channel no no no no
UnicastRTP Unicast RTP Media Channel Driver no no no no
MulticastRTP Multicast RTP Paging Channel Driver no no no no
PJSIP PJSIP Channel Driver yes no yes yes
Local Local Proxy Channel Driver yes no yes no
Surrogate Surrogate channel used to pull channel f no no no no
----------
9 channel drivers registered.
Do I have all the necessary modules loaded?
HostedTest1*CLI> module show like audiosocket
Module Description Use Count Status Support Level
app_audiosocket.so AudioSocket Application 0 Running extended
res_audiosocket.so AudioSocket support 1 Running extended
Are there any .conf files I’m supposed to be using to enable this functionality?