Ret
June 28, 2023, 10:50pm
1
Hi! I apologize if this topic has already been discussed before. I reall y couldn’t find a solution.
These are my settings for a trunk
type=endpoint
transport=0.0.0.0-udp
context=from-trunk
disallow=all
allow=alaw,ulaw,gsm
aors=4411998
send_connected_line=true
language=es_ar
user_eq_phone=no
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
trust_id_inbound=yes
t38_udptl_nat=no
send_pai=yes
rtp_symmetric=yes
dtmf_mode=rfc4733
I think there’s a problem in using PJSIP channels because with regular SIP it works perfectly.
Any ideas?
Because the prefix stripping is handled by FreePBX, the phone surely has no way of knowing that you’ve stripped the 9?
The phone is correctly displaying the number dialled.
david55
July 1, 2023, 8:19pm
3
In principle, the provider could provide a connected line update, and FreePBX might propagate that back to the phone. It would need trust connected line updates setting on the trunk, but that seems to be the Asterisk default, although FreePBX might override it.
Whether the provider does and how FreePBX, and the phone, handles it, are different questions.
Ret
July 3, 2023, 1:14pm
4
I tried adding that setting but it doesn´t strip the outbound route prefix.
My trunk is set to connect a Grandstream FXO. The route prefix is stripped if I use a SIP trunk but not a PJSIP trunk.
These are my trunk settings taken from asterisk CLI:
ParameterName : ParameterValue
================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (alaw|ulaw|gsm)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : MyTrunk
asymmetric_rtp_codec : false
auth :
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : <unknown>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : from-trunk
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : true
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 :
from_user :
g726_non_standard : false
ice_support : false
identify_by : username,ip
ignore_183_without_sdp : false
inband_progress : false
incoming_mwi_mailbox :
language : es_ar
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 : 0
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
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 : 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 : 0.0.0.0-udp
trust_connected_line : yes
trust_id_inbound : true
trust_id_outbound : true
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
system
August 3, 2023, 1:14pm
5
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