Asterisk Web RTC -Cant Answer Call

Asterisk Asterisk WebRTC
1

i have teseted 3- 4 Webrtc Phones (innivative Asterisk, Sara Phone etc) When the incoming call answer, it suddenly disconnected. My console showing following error.

WhatsApp Image 2024-05-02 at 14.46.14_52a2ef8f
WhatsApp Image 2024-05-02 at 14.46.14_52a2ef8f1204×826 163 KB

this is my sip.conf file

[general]
;register => 100:12345@172.20.8.54
;register => 200:12345@172.20.8.60

context = public
allowoverlap = no
tlsbindaddr = 0.0.0.0:8089
udpbindaddr = 0.0.0.0:5060
tcpenable = yes
tcpbindaddr = 0.0.0.0:5061
transport = udp,ws,wss,tls,tcp
srvlookup = yes
qualify = yes
realm = 0.0.0.0:5061
websocket_enabled=yes
tlsenable = yes
tlscertfile=/etc/asterisk/keys/dmscakey.pem
tlscafile=/etc/asterisk/keys/dmscakey.pem
icesupport=yes
nat=force_rport,comedia
dtlsenable=yes
webrtc=yes
dtlssetup=actpass
media_encryption=dtls
dtlsfingerprint=SHA-256
dtlscertfile=/etc/asterisk/keys/dmscakey.pem
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/dmscakey.pem
dtlscafile=/etc/asterisk/keys/dmscakey.pem
dtlssetup=actpass
allow=opus

[authentication]
basic-options
dtmfmode = rfc2833
context = from_external
type = friend
natted-phone
directmedia = no
host = dynamic
public-phone
directmedia = yes
my-codecs
disallow = all
allow = ilbc
allow = g729
allow = gsm
allow = g723
allow = ulaw
ulaw-phone
disallow = all
allow = ulaw

[100]
type = friend
host = dynamic
defaultuser = 100
dtmfmode=rfc2833
secret = 12345
fromuser = 100
fromdomain = 172.16.1.1
context = from_external
;disallow=all
;allow=opus,g722,ulaw,vp9,vp8,h264
;encryption=no
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=g722
allow=alaw
avpf=yes
icesupport=yes
directmedia=no
nat = force_rport,comedia
externip = 124.43.67.53
localnet = 172.20.10.0/24 ; Adjust this to match your local network subnet
qualify = yes
transport = ws, wss
encryption = yes
webrtc=yes
max_audio_streams = 1

[101]
type = friend
host = dynamic
defaultuser = 101
dtmfmode=rfc2833
secret = 12345
fromuser = 101
fromdomain = 172.16.1.1
context = from_external
;disallow=all
;allow=opus,g722,ulaw,vp9,vp8,h264
;encryption=no
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=g722
allow=alaw
avpf=yes
icesupport=yes
directmedia=no
nat = force_rport,comedia
externip = 124.43.67.53
localnet = 172.20.10.0/24 ; Adjust this to match your local network subnet
qualify = yes
transport = ws,wss
;encryption = yes
webrtc=yes
max_audio_streams = 1

How solve this problem???

Thank you.

2

This looks like chan_sip. I would start by updating your server, and using pjsip.

3 Likes
3

Thank you i will update accordigly and provide a feedback

4

We have made the necessary updates and can now receive calls. Thank you for your assistance. We appreciate it :heart_eyes:

5

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