I’m trying to set an extension for use over WSS. I could register it from -let’s say- https://tryit.jssip.net/ and I can even dial from or to it using another SIP extension in the same system.
The problem is I have no audio, and as the jssip app stops responding after I pick up, I think some kind of negotiation is not succeeding.
As there’s no official documentation on how to do this from FreePBX 17, I was following a bunch of howtos and articules from older versions, but it looks like I’m missing something.
Attached is how my registered endpoint looks like from the console. I wonder why “webrtc” is set to “no”, for instance. Do you see anything else that could be wrong?
Thank you very much.
asterisk -rx 'pjsip show endpoint 99000'
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.........................................................>
Match: <criteria.........................>
Channel: <ChannelId......................................> <State.....> <Time.....>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
==========================================================================================
Endpoint: 99000/99000 Not in use 0 of inf
InAuth: 99000-auth/99000
Aor: 99000 1
Contact: 99000/sip:[email protected]:36475;tr c3a336f306 Avail 28.265
Transport: 0.0.0.0-wss wss 3 96 0.0.0.0:5060
ParameterName : ParameterValue
===================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (ulaw|alaw|g729|gsm|opus|g726|g722)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : 99000
asymmetric_rtp_codec : false
auth : 99000-auth
bind_rtp_to_media_address : false
bundle : true
call_group :
callerid : "My Extension" <99000>
callerid_privacy : allowed_not_screened
callerid_tag :
codec_prefs_incoming_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_incoming_offer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_offer : prefer:pending, operation:union, keep:all, transcode:allow
connected_line_method : invite
contact_acl :
context : from-internal
cos_audio : 5
cos_video : 4
device_state_busy_at : 0
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_auto_generate_cert : Yes
dtls_ca_file :
dtls_ca_path :
dtls_cert_file :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : actpass
dtls_verify : No
dtmf_mode : rfc4733
fax_detect : false
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
geoloc_incoming_call_profile :
geoloc_outgoing_call_profile :
ice_support : true
identify_by : username,ip
ignore_183_without_sdp : false
inband_progress : false
incoming_call_offer_pref : local
incoming_mwi_mailbox :
language : es_AR
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : dtls
media_encryption_optimistic : false
media_use_received_transport : true
message_context :
moh_passthrough : false
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : no
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : true
outbound_auth :
outbound_proxy :
outgoing_call_offer_pref : remote_merge
overlap_context :
pickup_group :
preferred_codec_only : false
record_off_feature : apprecord
record_on_feature : apprecord
redirect_method : user
refer_blind_progress : true
rewrite_contact : true
rpid_immediate : false
rtcp_mux : true
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 30
rtp_timeout_hold : 300
sdp_owner : -
sdp_session : Asterisk
security_negotiation : no
send_aoc : false
send_connected_line : yes
send_diversion : true
send_history_info : false
send_pai : true
send_rpid : false
set_var :
srtp_tag_32 : false
stir_shaken : no
stir_shaken_profile :
sub_min_expiry : 0
subscribe_context :
suppress_moh_on_sendonly : false
suppress_q850_reason_headers : false
t38_bind_udptl_to_media_address : false
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
tenantid :
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 184
tos_video : 136
transport : 0.0.0.0-wss
trust_connected_line : yes
trust_id_inbound : true
trust_id_outbound : false
use_avpf : true
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no