Trying to setup Asterisk for voice chat between website users with sipjs. But unable to configure DTLS certificates. getting "hostname: Unknown host"

Question

I'm trying to setup Asterisk Voice chat for users with the Help of Sipjs follows the instruction given on SIPJS docs http://sipjs.com/guides/server-configuration/asterisk. Users are created and also connected. They can call each other through Zoiper. But Unable to Call through Sipjs or SipML5. When anyone user calls from Sipjs or SipMl5. Console shows following error :

Connected to Asterisk 11.20.0 currently running on asterix (pid = 13719)
[Oct 14 05:25:22] NOTICE[13735][C-00000000]: chan_sip.c:25844 handle_request_invite: Call from '' (88.150.240.102:5071) to extension '90041215085741' rejected because extension not found in context 'default'.
[Oct 14 05:25:46] NOTICE[13735][C-00000001]: chan_sip.c:10005 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 23496 UDP/TLS/RTP/SAVPF 109 9 0 8
[Oct 14 05:25:46] WARNING[13735][C-00000001]: chan_sip.c:10398 process_sdp: Rejecting secure audio stream without encryption details: audio 23496 UDP/TLS/RTP/SAVPF 109 9 0 8
[Oct 14 05:25:54] WARNING[13735]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 1faf349623b90d4f62fe562ae66d6c45 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Oct 14 05:25:55] NOTICE[13735][C-00000002]: chan_sip.c:25844 handle_request_invite: Call from '' (88.150.240.102:5070) to extension '0041215085741' rejected because extension not found in context 'default'.

and also during installation of DTLS certificates I'm getting "hostname: Unknown host". Anybody please guide me how to setup Asterisk Voice chat correctly.

Answer 1

From what i see in your console - 1st - the extension dose not exist in you context 2nd - set "avpf=yes" in your sip user config 3rd - set "encryption=yes" in your sip user config

From your title what i get to know is that, you have not generated certificates for that -

mkdir /etc/asterisk/keys
cd /usr/src/asterisk-13.1.0/contrib/scripts
/ast_tls_cert -C hostname -O "Some Name" -d /etc/asterisk/keys

here is an example for sip config for webphone phone:

[1000]
host=dynamic
secret=YouPassword
context=from-internal
type=friend
encryption=yes
avpf=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
allow=alaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

If you still have problem then share me your configurations(hope you u have configured webrtc support)