Taken from previous threads: "
You cannot use the same endpoint for a phone and webrtc*
".
I have defined two endpoints in pjsip, one for a softphone phone and the other for webrtc.
I can make calls from webrtc to a softphone but not vice-versa.
Is call initiation from both endpoints in all ways possible?
If configured correctly, yes. You haven’t provided the configuration, or stated what actually happens when it doesn’t work, or provided logging so noone can help any further with that aspect.
Thank you for your swift response, i get the following error:
pp_dial.c:3381 dial_exec_full: Had to drop call because I couldn’t make PJSIP/100-00000004 compatible with PJSIP/200-00000005
200 - softphone
100 - webrtc phone
Below are my configurations
;pjsip.conf
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
local_net = 0.0.0.0/24
allow_reload=true
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
;webrtc=yes
; All other transport parameters are ignored for wss transports.
[100]
type=endpoint
context=internal
disallow=all
allow=opus,ulaw
auth=100
aors=100
dtls_auto_generate_cert=yes
webrtc=yes
[100]
type=auth
auth_type=userpass
username=100
password=100
[100]
type=aor
max_contacts=2
[200]
type=endpoint
context=internal
disallow=all
allow=opus,ulaw
auth=200
aors=200
dtls_auto_generate_cert=yes
;webrtc=yes
[200]
type=auth
auth_type=userpass
username=200
password=200
[200]
type=aor
max_contacts=2
That would mean each side is using different codecs, and there is no translation path between them. Likely one side is opus and the other isn’t, and you don’t have a codec_opus to transcode between the two. You could confirm this by removing opus from the allow list on both, or selecting codec_opus in menuselect when installing Asterisk.
I removed opus and its now now working, thank you 
This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.