Issue with inbound calls using freepbx 17 pjsipe

fraddygil · September 4, 2024, 8:40am

Hi, guys,

I have a freepbx setup with a Goip “sim gateeway”.

I am having problems getting calls into my freepbx.

I can make calls without problems through the trunk but when I get them to return to asterisk. I get the following messages back.

I leave logs.

Can you help me?

Asterisk 21.4.1, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 21.4.1 currently running on localhost (pid = 112941)
  == Endpoint goip11 is now Unreachable
    -- Contact goip11/sip:[email protected]:5288 is now Unreachable.  RTT: 0.000 msec
    -- Removed contact 'sip:[email protected]:5288' from AOR 'goip11' due to expiration
  == Contact goip11/sip:[email protected]:5288 has been deleted
    -- Added contact 'sip:[email protected]:5288' to AOR 'goip11' with expiration of 60 seconds
  == Endpoint goip11 is now Reachable
    -- Contact goip11/sip:[email protected]:5288 is now Reachable.  RTT: 59.375 msec
[2024-09-03 23:18:44] NOTICE[138804]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - No matching endpoint found
[2024-09-03 23:18:44] NOTICE[286519]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - No matching endpoint found
[2024-09-03 23:18:44] NOTICE[286519]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - Failed to authenticate
[2024-09-03 23:18:44] NOTICE[578263]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - No matching endpoint found
[2024-09-03 23:18:44] NOTICE[578263]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - Failed to authenticate
[2024-09-03 23:18:44] NOTICE[127114]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - No matching endpoint found
[2024-09-03 23:18:44] NOTICE[127114]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - Failed to authenticate
[2024-09-03 23:18:45] NOTICE[132141]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - No matching endpoint found
[2024-09-03 23:18:45] NOTICE[132141]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - Failed to authenticate
[2024-09-03 23:18:45] NOTICE[138804]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - No matching endpoint found
[2024-09-03 23:18:45] NOTICE[138804]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'INVITE' from '"+18143778645" <sip:[email protected]>' failed for '192.168.1.240:5288' (callid: [email protected]) - Failed to authenticate
    -- Removed contact 'sip:[email protected]:47656;ob;x-ast-orig-host=10.100.1.96:47656' from AOR '6000' due to request
  == Contact 6000/sip:[email protected]:47656;ob;x-ast-orig-host=10.100.1.96:47656 has been deleted
  == Endpoint 6000 is now Unreachable
    -- Added contact 'sip:[email protected]:47656;ob;x-ast-orig-host=10.100.1.96:47656' to AOR '6000' with expiration of 300 seconds
  == Endpoint 6000 is now Reachable
    -- Contact 6000/sip:[email protected]:47656;ob;x-ast-orig-host=10.100.1.96:47656 is now Reachable.  RTT: 129.734 msec

TRUNK:

root@localhost:/etc/asterisk# cat cat /etc/asterisk/pjsip.endpoint.conf
cat: cat: No such file or directory
;--------------------------------------------------------------------------------;
;          Do NOT edit this file as it is auto-generated by FreePBX.             ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate.                                                         ;
;--------------------------------------------------------------------------------;

[0]
#include pjsip.endpoint_custom.conf
[6000]
type=endpoint
aors=6000
auth=6000-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,gsm,g726,g722
context=from-internal
callerid=MIT <6000>

dtmf_mode=rfc4733
direct_media=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
user_eq_phone=no
send_connected_line=yes
media_encryption=no
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=yes
rtp_timeout=30
rtp_timeout_hold=300
rtp_keepalive=0
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord

[101]
type=endpoint
aors=101
auth=101-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,gsm,g726,g722
context=from-internal
callerid=101 <101>

dtmf_mode=rfc4733
direct_media=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
user_eq_phone=no
send_connected_line=yes
media_encryption=no
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=yes
rtp_timeout=30
rtp_timeout_hold=300
rtp_keepalive=0
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord

[102]
type=endpoint
aors=102
auth=102-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,gsm,g726,g722
context=from-internal
callerid=102 <102>

dtmf_mode=rfc4733
direct_media=yes
mailboxes=102@default

mwi_subscribe_replaces_unsolicited=yes
aggregate_mwi=no
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
user_eq_phone=no
send_connected_line=no
media_encryption=no
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=yes
rtp_timeout=30
rtp_timeout_hold=300
rtp_keepalive=0
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord

[goip11]
type=endpoint
transport=0.0.0.0-udp
context=from-pstn
disallow=all
allow=ulaw,alaw,gsm,g726,g722,g719,h264,mpeg4
aors=goip11
send_connected_line=no
rtp_keepalive=0
language=en
outbound_auth=goip11
user_eq_phone=no
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
trust_id_inbound=no
t38_udptl_nat=no
direct_media=no
rtp_symmetric=yes
dtmf_mode=auto

[goip12]
type=endpoint
transport=0.0.0.0-udp
context=from-pstn
disallow=all
allow=ulaw,alaw,gsm,g726,g722,g719,h264,mpeg4
aors=goip12
send_connected_line=no
rtp_keepalive=0
language=en
outbound_auth=goip12
user_eq_phone=no
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
trust_id_inbound=no
t38_udptl_nat=no
direct_media=no
rtp_symmetric=yes
dtmf_mode=auto

[dpma_endpoint]
type=endpoint
context=dpma-invalid

tmittelstaedt · September 5, 2024, 2:21pm

If you look at Appendix B of the user manual for that device you will see a complete set of SIP INVITE handshakes, and so on.

What you are going to need to do is at the Debian command line do:

sudo apt install wireshark

and then run wireshark while attempting to use the gateway and capture the SIP transaction. It should then be a simple matter to match the output of wireshark against what the goIP gateway says is supposed to be happening then see what any discrepancies might be.

dicko · September 5, 2024, 2:44pm

Fire up sngrep and filter for from IP 192.168.1.240

fraddygil · September 6, 2024, 5:33pm

I have solved it in the pjsip trunk. indicating that for incoming validate authentication

Thanks guys

system · October 7, 2024, 5:34pm

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