res_pjsip_registrar.c: Endpoint 'anonymous' has no configured AORs #412

Adportas · 2023-01-24T20:03:36Z

Adportas
Jan 24, 2023

Hi,
I followed that tutorial: https://vitalpbx.com/blog/asterisk-webrtc-desde-cero/
Based on this one: https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients
It uses InnovateAsterisk Browser-Phone but when i try to register it allways claim i'am using anonymous ids rather than the ones i put in the web interface. I am a bit confused as to where the problem might be, could you please tell me where to check?

thank you in advance
Daniel

Adportas · 2023-01-24T20:05:35Z

Adportas
Jan 24, 2023
Author

0 replies

InnovateAsterisk · 2023-01-25T06:09:18Z

InnovateAsterisk
Jan 25, 2023
Maintainer

Can you post a copy of the relevant part of the config file. This message indicates that you have not set any aor entry in the endpoint configuration in vital pbx (asterisk side)

1 reply

Adportas Jan 25, 2023
Author

Hi,
Thank you for answering, here are the modified configurations to make it work, anything missing to diagnose please let me know and I will add it.
Thank you very much
Daniel

We disable websocket for sip:

[root@localhost asterisk]# cat sip_general_custom.conf

language=pt_BR
websocket_enabled=false

The users ...

[root@localhost asterisk]# cat pjsip.auth_custom.conf

;[1300-auth]
;type=auth
;auth_type=userpass
;password=ADP1300
;username=1300

[webrtc-phones](!)
context=main-context
transport=transport-wss
allow=!all,opus,ulaw,alaw,vp8,vp9
webrtc=yes

[User1](webrtc-phones)
type=endpoint
callerid=”User One” <100>
auth=User1
aors=User1

[User1]
type=aor
max_contacts=3

[User1]
type=auth
auth_type=userpass
username=User1
password=1234

[User2](webrtc-phones)
type=endpoint
callerid=”User Two” <101>
auth=User2
aors=User2

[User2]
type=aor
max_contacts=3

[User2]
type=auth
auth_type=userpass
username=User2
password=1234

[User3](webrtc-phones)
type=endpoint
callerid=”User Three” <102>
auth=User3
aors=User3

[User3]
type=aor
max_contacts=3

[User3]
type=auth
auth_type=userpass
username=User3
password=1234

We add some required configurations in custom:

[root@localhost asterisk]# cat pjsip_custom.conf

[1300]
type=endpoint
aors=1300
auth=1300-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,g722
context=from-internal
callerid=Francisco <1300>
;Added
dtmf_mode=rfc4733
direct_media=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=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
media_encryption_optimistic=no
rtp_timeout=30
rtp_timeout_hold=300
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=es
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord

The transport

[root@localhost asterisk]# cat pjsip.transports_custom.conf

[system]
type=system
timer_t1=500
timer_b=32000
disable_tcp_switch=yes

[global]
type=global
max_initial_qualify_time=0
keep_alive_interval=90
contact_expiration_check_interval=30
default_voicemail_extension=*97
unidentified_request_count=3
unidentified_request_period=5
unidentified_request_prune_interval=30
mwi_tps_queue_high=500
mwi_tps_queue_low=-1
mwi_disable_initial_unsolicited=yes
send_contact_status_on_update_registration=yes

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089
local_net=192.168.100.0/24
external_media_address=10.10.10.4
external_signaling_address=10.10.10.4
allow_reload=yes

The extensions file in custom

[root@localhost asterisk]# cat extensions_custom.conf

exten => 1300,1,Answer
exten => 1300,n,AGI(agi://192.168.100.17/helo.agi)

include => app-custom-chanspy

[app-custom-chanspy]
exten => _550.,1,Macro(user-callerid,)
exten => _550.,n,ChanSpy(SIP/${EXTEN:3},q)
exten => _550.,n,Hangup

exten => _551.,1,Macro(user-callerid,)
exten => _551.,n,ChanSpy(SIP/${EXTEN:3},w)
exten => _551.,n,Hangup

exten => _552.,1,Macro(user-callerid,)
exten => _552.,n,ChanSpy(SIP/${EXTEN:3},B)
exten => _552.,n,Hangup


[general]
static=yes
writeprotect=yes
priorityjumping=no
autofallthrough=no

[globals]
ATTENDED_TRANSFER_COMPLETE_SOUND=beep

[main-context]
include => from-extensions
include => subscriptions
include => textmessages
include => echo-test
include => speak-exte-nnum

[echo-test]
exten => 777,1,NoOp(FEATURE: ECHO TEST)
 same => n,Answer
 same => n,Wait(1)
 same => n,Playback(demo-echotest)
 same => n,Echo()
 same => n,Playback(demo-echodone)
 same => n,Hangup()
;END of [echo-test]

[speak-exte-nnum]
exten => 888,1,NoOp(FEATURE: SPEAK MY EXTENSION NUMBER)
 same => n,Answer
 same => n,Wait(1)
 same => n,Playback(extension)
 same => n,Wait(1)
 same => n,SayDigits(${CALLERID(num)})
 same => n,Wait(2)
 same => n,Hangup()
;END of [speak-exte-nnum]

[textmessages]
exten => 100,1,Gosub(send-text,s,1,(User1))
exten => 101,1,Gosub(send-text,s,1,(User2))
exten => 102,1,Gosub(send-text,s,1,(User3))

[subscriptions]
exten => 100,hint,PJSIP/User1
exten => 101,hint,PJSIP/User2
exten => 102,hint,PJSIP/User3

[from-extensions]
; Feature Codes:
exten => *65,1,Gosub(moh,s,1)
; Extensions
exten => 100,1,Gosub(dial-extension,s,1,(User1))
exten => 101,1,Gosub(dial-extension,s,1,(User2))
exten => 102,1,Gosub(dial-extension,s,1,(User3))

exten => e,1,Hangup()

[moh]
exten => s,1,NoOp(Music On Hold)
exten => s,n,Ringing()
exten => s,n,Wait(2)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,MusicOnHold()

[dial-extension]
exten => s,1,NoOp(Calling: ${ARG1})
exten => s,n,Set(JITTERBUFFER(adaptive)=default)
exten => s,n,Dial(PJSIP/${ARG1},30)
exten => s,n,Hangup()

exten => e,1,Hangup()

[send-text]
exten => s,1,NoOp(Sending Text To: ${ARG1})
exten => s,n,Set(PEER=${CUT(CUT(CUT(MESSAGE(from),@,1),<,2),:,2)})
exten => s,n,Set(FROM=${SHELL(asterisk -rx ‘pjsip show endpoint ${PEER}’ | grep ‘callerid ‘ | cut -d’:’ -f2- | sed ‘s/^ *//’ | tr -d ‘
‘)})
exten => s,n,Set(CALLERID_NUM=${CUT(CUT(FROM,>,1),<,2)})
exten => s,n,Set(FROM_SIP=${STRREPLACE(MESSAGE(from),
exten => s,n,MessageSend(pjsip:${ARG1},${FROM_SIP})
exten => s,n,Hangup()

InnovateAsterisk · 2023-01-25T06:12:52Z

InnovateAsterisk
Jan 25, 2023
Maintainer

https://github.com/InnovateAsterisk/Browser-Phone#chan_pjsip

0 replies

InnovateAsterisk · 2023-01-26T10:01:34Z

InnovateAsterisk
Jan 26, 2023
Maintainer

So somehow its picking up the account as anonymous. But If i look at that this:

[User1](webrtc-phones)
type=endpoint
callerid=”User One” <100>
auth=User1
aors=User1

[User1]
type=aor
max_contacts=3

[User1]
type=auth
auth_type=userpass
username=User1
password=1234

User1 is has clearly been configured.

First do this lets do some sanity tests. reload the Asterisk service. Then go into the CLI and issue:

CLI> pjsip show endpoints

Make sure that the output looks something like this:

 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:  User1/100                     Unavailable   0 of 1
     InAuth:  User1/User1
        Aor:  User1                                        10

It could be a simple as the dial plan reader not able to parse the characters in the line callerid=”User Two” <101> (you will see here the ” is not a ").

2 replies

Adportas Jan 26, 2023
Author

Hi
This is what the command displays
Thankful
Daniel

localhost*CLI> pjsip show endpoints

 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:  1300/1300                                            Unavailable   0 of inf
        Aor:  1300                                               1

 Endpoint:  201/201                                              Unavailable   0 of inf
     InAuth:  201-auth/201
        Aor:  201                                                1

 Endpoint:  205/205                                              Not in use    0 of inf
     InAuth:  205-auth/205
        Aor:  205                                                1
      Contact:  205/sip:[email protected]:53877;ob       6be20c50c8 Avail         1.887

 Endpoint:  210/210                                              Unavailable   0 of inf
     InAuth:  210-auth/210
        Aor:  210                                                1

 Endpoint:  300/300                                              Unavailable   0 of inf
     InAuth:  300-auth/300
        Aor:  300                                                1

 Endpoint:  a_CUCM_pjsip                                         Not in use    0 of inf
        Aor:  a_CUCM_pjsip                                       0
      Contact:  a_CUCM_pjsip/sip:10.10.10.4:5060           64161c5f4d Avail         3.103
  Transport:  0.0.0.0-udp               udp      3     96  0.0.0.0:5060
   Identify:  a_CUCM_pjsip/a_CUCM_pjsip
        Match: 10.10.10.4/32

 Endpoint:  anonymous                                            Unavailable   0 of inf


Objects found: 7

Adportas Jan 26, 2023
Author

Hi
The extensions defined in the pjsip.auth_custom.conf file are being ignored, while the ones defined via freepbx are the only ones shown by the pjsip show endpoints command, so we tried with one of those and the authentication worked.
Thank you very much for your support
Daniel

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res_pjsip_registrar.c: Endpoint 'anonymous' has no configured AORs #412

Adportas Jan 24, 2023

Replies: 4 comments · 3 replies

Adportas Jan 24, 2023 Author

InnovateAsterisk Jan 25, 2023 Maintainer

Adportas Jan 25, 2023 Author

We disable websocket for sip:

The users ...

We add some required configurations in custom:

The transport

The extensions file in custom

InnovateAsterisk Jan 25, 2023 Maintainer

InnovateAsterisk Jan 26, 2023 Maintainer

Adportas Jan 26, 2023 Author

localhost*CLI> pjsip show endpoints

Adportas Jan 26, 2023 Author

Adportas
Jan 24, 2023

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Adportas
Jan 24, 2023
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InnovateAsterisk
Jan 25, 2023
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Adportas Jan 25, 2023
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InnovateAsterisk
Jan 25, 2023
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InnovateAsterisk
Jan 26, 2023
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Adportas Jan 26, 2023
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Adportas Jan 26, 2023
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