Hi,
After spending an hour on the phone with my providers, it seems that I have a problem with my settings for my endpoint or my transport. Where ? we did not find…
So we have a concern for one way audio.
If I call an external number from my asterisk, the person hears me, but I cannot hear the person.
Config :
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.13.0/24
external_media_address=82.212.***.**
external_signaling_address=82.212.***.**
[*****]
type = registration
transport=transport-udp
server_uri = sip:sip.***.be
client_uri = sip:77*******@sip.***.be
outbound_auth = AUTH
auth_rejection_permanent = no
retry_interval = 30
forbidden_retry_interval = 300
max_retries = 20
contact_user = +3242960077
Endpoint: ****/+3242960077 Not in use 0 of inf
OutAuth: *****/778******
InAuth: *****/77********
Aor: ***** 10
Contact: *****/sip:sip.****.be 6c70569b3a Avail 33.323
Transport: transport-udp udp 0 0 0.0.0.0:5060
ParameterName : ParameterValue
=========================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (ulaw|alaw)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : ****
asymmetric_rtp_codec : false
auth : ******
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : +3242960077
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
contact_user : +3242960077
context : appel_in
cos_audio : 0
cos_video : 0
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 : No
dtls_ca_file :
dtls_ca_path :
dtls_cert_file :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : active
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 : sip.*****.be
from_user : 77********
g726_non_standard : false
ice_support : false
identify_by : username,ip
ignore_183_without_sdp : false
inband_progress : false
incoming_mwi_mailbox :
language :
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
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 : false
outbound_auth : ******
outbound_proxy :
pickup_group :
preferred_codec_only : false
record_off_feature : automixmon
record_on_feature : automixmon
refer_blind_progress : true
rewrite_contact : false
rpid_immediate : false
rtcp_mux : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 1
rtp_symmetric : false
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_connected_line : yes
send_diversion : true
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 0
subscribe_context :
suppress_q850_reason_headers : false
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport : transport-udp
trust_connected_line : yes
trust_id_inbound : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
I have a simple auth with userpass and an aors.
So here are the asterisk logs when I call a number :
Executing [+32496424040@appel_internes_lgtech:2] Dial("PJSIP/11202-00000004", "PJSIP/+32496424040@SIPIT") in new stack
-- Called PJSIP/+3249********@******
-- PJSIP/*****-00000005 is ringing
-- PJSIP/*****-00000005 is ringing
> 0x7f667c06f4c0 -- Strict RTP learning after remote address set to: 109.**.**.**.**:15940
-- PJSIP/*****-00000005 answered PJSIP/11202-00000004
> 0x7f667c149fb0 -- Strict RTP learning after remote address set to: 192.168.13.44:12620
-- Channel PJSIP/*****-00000005 joined 'simple_bridge' basic-bridge <0cd916af-6963-44ba-8b4d-3347ecdd5265>
-- Channel PJSIP/11202-00000004 joined 'simple_bridge' basic-bridge <0cd916af-6963-44ba-8b4d-3347ecdd5265>
> 0x7f667c149fb0 -- Strict RTP switching to RTP target address 192.168.13.44:12620 as source
> 0x7f667c149fb0 -- Strict RTP learning complete - Locking on source address 192.168.13.44:12620
-- Channel PJSIP/11202-00000004 left 'simple_bridge' basic-bridge <0cd916af-6963-44ba-8b4d-3347ecdd5265>
== Spawn extension (appel_internes_lgtech, +32496424040, 2) exited non-zero on 'PJSIP/11202-00000004'
-- Executing [h@appel_internes_lgtech:1] Goto("PJSIP/11202-00000004", "Hangup,s,1") in new stack
-- Goto (Hangup,s,1)
-- Executing [s@Hangup:1] NoOp("PJSIP/11202-00000004", ""+3242****** A RACROCHE"") in new stack
-- Executing [s@Hangup:2] Hangup("PJSIP/11202-00000004", "") in new stack
== Spawn extension (Hangup, s, 2) exited non-zero on 'PJSIP/11202-00000004'
-- Channel PJSIP/******-00000005 left 'simple_bridge' basic-bridge <0cd916af-6963-44ba-8b4d-3347ecdd5265>
Sorry for the ‘*’ My providers are very careful not to reveal anything on the internet