another issue -
with my matrix Gateway…
on chansip all is working fine but on pjsip outgoing calls are going fine…
but incoming calls are not landing on the freepbx…
asterisk log
matrix sngrep log
my chansip settings
type=peer
qualify=yes
port=5060
nat=yes
insecure=very
host=192.168.1.240
context=from-internal
sip and pjsip listen on different ports. You are stating that pjsip is not working but you only posted the sip configuration. Make sure your calls are being sent to the proper UDP port for pjsip.
is there a command for CLI from where i can get the pjsip conf settings for the trunk?
it is very inconvenient to take screenshots and post them here and then experts like you to read it.
The log in your penultimate post shows a complete call, but there was an attempt on line 717 to call 31 at BSNL, which of course was rejected. The call was then sent to the Matrix and accepted. I did not attempt to analyze why this happened, but BSNL might block your account if you send them too many invalid requests.
The Asterisk log in your last post only shows an outbound call via the Matrix, which appeared to be completely successful. If something was wrong with it, please explain.
The ‘sngrep’ log in your last post shows Asterisk requesting authentication from the Matrix, which it apparently didn’t provide. Try configuring the pjsip trunk for Authentication None. If it’s already set that way, put 192.168.1.240 in Match (Permit) for the trunk. If you still have trouble, paste the Asterisk log for such a call, including pjsip logger.
Screenshots or your trunk settings, both the General and Advanced tabs of pjsip Settings, should be fine.
Or, from the Asterisk command prompt:
pjsip show endpoint matrix_pjsip
pjsip show auth matrix_pjsip
pjsip show aor matrix_pjsip
pjsip show identify matrix_pjsip
pjsip show registration matrix_pjsip
mnhpbx*CLI> pjsip show endpoint matrix_pjsip
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: matrix_pjsip Invalid 0 of inf
Aor: matrix_pjsip 0
Contact: matrix_pjsip/sip:192.168.1.240:5060 e7d1d1462e NonQual nan
Transport: 0.0.0.0-udp udp 3 96 0.0.0.0:5160
Identify: matrix_pjsip/matrix_pjsip
Match: 192.168.1.240/32
ParameterName : ParameterValue
===================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (ulaw|alaw|gsm|g726|g722|h265|h264|mpeg4)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : matrix_pjsip
asymmetric_rtp_codec : false
auth :
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : <unknown>
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 : 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 : auto
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 : false
identify_by : username,ip
ignore_183_without_sdp : false
inband_progress : false
incoming_call_offer_pref : local
incoming_mwi_mailbox :
language : en
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 :
outgoing_call_offer_pref : remote_merge
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_aoc : false
send_connected_line : no
send_diversion : true
send_history_info : false
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
stir_shaken : off
stir_shaken_profile :
sub_min_expiry : 0
subscribe_context :
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
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 : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
mnhpbx*CLI> pjsip show auth matrix_pjsip
I/OAuth: <AuthId/UserName.............................................................>
==========================================================================================
Auth: matrix_pjsip/matrix_pjsip
ParameterName : ParameterValue
===============================
auth_type : userpass
md5_cred :
nonce_lifetime : 32
oauth_clientid :
oauth_secret :
password :
realm :
refresh_token :
username : matrix_pjsip
mnhpbx*CLI> pjsip show aor matrix_pjsip
Aor: <Aor..............................................> <MaxContact>
Contact: <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================
Aor: matrix_pjsip 0
Contact: matrix_pjsip/sip:192.168.1.240:5060 e7d1d1462e Avail 13.027
ParameterName : ParameterValue
=============================================
authenticate_qualify : false
contact : sip:192.168.1.240:5060
default_expiration : 3600
mailboxes :
max_contacts : 0
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy :
qualify_frequency : 60
qualify_timeout : 3.000000
remove_existing : false
remove_unavailable : false
support_path : false
voicemail_extension :
mnhpbx*CLI> pjsip show identify matrix_pjsip
Identify: <Identify/Endpoint...........................................................>
Match: <criteria...........................>
==========================================================================================
Identify: matrix_pjsip/matrix_pjsip
Match: 192.168.1.240/32
ParameterName : ParameterValue
=============================================
endpoint : matrix_pjsip
match : 192.168.1.240/255.255.255.255
match_header :
srv_lookups : true
mnhpbx*CLI> pjsip show registration matrix_pjsip
Unable to find object matrix_pjsip.
I’m puzzled. If both FreePBX and Matrix are statically configured (neither side is registering to the other) and you have Authentication None, I don’t understand why Asterisk is trying to auth the INVITE from the Matrix.
Paste the Asterisk log (with pjsip logger on) for a failing call from Matrix to FreePBX.
Sorry, I missed what was wrong:
You have pjsip Port to Listen On = 5160 and chan_sip Bind Port =5060. That’s backwards from the default settings, but is not a problem.
Then, you have the Matrix configured to send calls to port 5060 on FreePBX, so it’s going to chan_sip, where there is no trunk, so a phony auth request gets dummied up, and the Matrix chokes.
Configure the Matrix to send these calls to port 5160 and you should be good to go.
ok yeah … thats correct…
another thing…
is it possible to have all FXO ports on the matrix register as individual trunks rather than peer-2-peer connection as a single trunk… the rotation of the trunks is done at matrix end…also i can select specific trunk using specific prefix when dialing and matrix removes that prefix at its end…
Just on a side note…is it significantly better to use pjsip instead on chansip??
i personally find chansip much easier to configure and is more forgiving…
i changed the port on matrix as well as on freepbx trunk side…
i am able to make outbound calls but inbound calls are still not not authorised
You don’t have to change anything on the PBX side if you changed Matrix to send UDP traffic to port 5160 on your end.
Is there a firewall that needs to be configured to port forward the correct ports for inbound traffic?
sorry for the delay in replying … it is still not working…
i will post the changes with pictures after the working hours to avoid any disconnections.
For your logs, do both
pjsip set logger on
and
sip set debug on
so we can see if any traffic is going (incorrectly) to chan_sip.
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