Find_registrar_aor: AOR '' not found for endpoint '6004'

Hello. I’m trying to make outbound call using Browser Phone software. When I’m going to register new account, following error showing asterisk CLI, How fix it?

  == WebSocket connection from '172.20.8.45:10772' closed
  == WebSocket connection from '172.20.8.45:10973' for protocol 'sip' accepted using version '13'
[May 20 14:06:58] WARNING[5666]: res_pjsip_registrar.c:1166 find_registrar_aor: AOR '' not found for endpoint '6004' (172.20.8.45:10973)
dms*CLI>

And this is my pjsip.conf confihurations

[transport-ws]
type=transport
protocol=ws ; Use `wss` for WebSocket Secure
bind=0.0.0.0:8088 ; Adjust the IP and port as necessary

[transport-wss]
type=transport
protocol=wss ; WebSocket Secure
bind=0.0.0.0:8089 
cert_file=/etc/asterisk/keys/asterisk.pem ; SSL certificate file
priv_key_file=/etc/asterisk/keys/asterisk.pem ; SSL private key file

[6004]
type=endpoint
transport=transport-wss
context=from_external
disallow=all
allow=opus
allow=ulaw
aors=6004_aor  
auth=6004_auth
media_encryption=dtls
dtls_auto_generate_cert=yes
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
direct_media=no
force_rport=yes
rewrite_contact=yes
webrtc=yes


[6004_auth]
type=auth
auth_type=userpass
username=6004
password=6004

[6004_aor]
type=aor
max_contacts=5
remove_existing=yes
contact=sip:6004@172.20.10.100

[provider_6004]
type=registration
outbound_auth=provider_6004_auth
server_uri=sip:172.16.1.1
client_uri=sip:6004@172.16.1.1
contact_user=6004
retry_interval=60
max_retries=10
transport=transport-wss

[provider_6004_auth]
type=auth
auth_type=userpass
username=6004
password=6004


Thank you

The user part of the To header, in the REGISTER request appears to be empty. Using pjsip set logger on should confirm this.

Hello, I add this part above in my code.

register => 01120xxxxx:*@172.16.1.1:5060

This is my pjsip log

PJSIP Logging enabled
  == WebSocket connection from '172.20.8.45:4722' closed
  == WebSocket connection from '172.20.8.45:4724' for protocol 'sip' accepted using version '13'
<--- Received SIP request (552 bytes) from WSS:172.20.8.45:4724 --->
REGISTER sip:172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.199;branch=z9hG4bK9310015
To: <sip:6004@172.20.10.100>
From: "6004" <sip:6004@172.20.10.100>;tag=4u40efpi1b
CSeq: 2 REGISTER
Call-ID: rtb9q3h4334i8mlfcok5
Max-Forwards: 70
Contact: <sip:ivkjgjdl@192.0.2.199;transport=wss>;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:125.0) Gecko/20100101 Firefox/125.0
Content-Length: 0


[May 20 16:22:12] NOTICE[6074]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '"6004" <sip:6004@172.20.10.100>' failed for '172.20.8.45:4724' (callid: rtb9q3h4334i8mlfcok5) - No matching endpoint found
<--- Transmitting SIP response (459 bytes) to WSS:172.20.8.45:4724 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.199;rport=4724;received=172.20.8.45;branch=z9hG4bK9310015
Call-ID: rtb9q3h4334i8mlfcok5
From: "6004" <sip:6004@172.20.10.100>;tag=4u40efpi1b
To: <sip:6004@172.20.10.100>;tag=z9hG4bK9310015
CSeq: 2 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1716202332/67ef07157b87d51e5460cd66c93211be",opaque="6e11bef56a5699da",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (820 bytes) from WSS:172.20.8.45:4724 --->
REGISTER sip:172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.199;branch=z9hG4bK3484509
To: <sip:6004@172.20.10.100>
From: "6004" <sip:6004@172.20.10.100>;tag=4u40efpi1b
CSeq: 3 REGISTER
Call-ID: rtb9q3h4334i8mlfcok5
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="6004", realm="asterisk", nonce="1716202332/67ef07157b87d51e5460cd66c93211be", uri="sip:172.20.10.100", response="dd3abdb919e80d32e9035254f4ab9782", opaque="6e11bef56a5699da", qop=auth, cnonce="o53lvf9g0dg1", nc=00000001
Contact: <sip:ivkjgjdl@192.0.2.199;transport=wss>;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:125.0) Gecko/20100101 Firefox/125.0
Content-Length: 0


[May 20 16:22:12] NOTICE[6075]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '"6004" <sip:6004@172.20.10.100>' failed for '172.20.8.45:4724' (callid: rtb9q3h4334i8mlfcok5) - No matching endpoint found
[May 20 16:22:12] NOTICE[6075]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '"6004" <sip:6004@172.20.10.100>' failed for '172.20.8.45:4724' (callid: rtb9q3h4334i8mlfcok5) - Failed to authenticate
<--- Transmitting SIP response (459 bytes) to WSS:172.20.8.45:4724 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.199;rport=4724;received=172.20.8.45;branch=z9hG4bK3484509
Call-ID: rtb9q3h4334i8mlfcok5
From: "6004" <sip:6004@172.20.10.100>;tag=4u40efpi1b
To: <sip:6004@172.20.10.100>;tag=z9hG4bK3484509
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1716202332/67ef07157b87d51e5460cd66c93211be",opaque="2deb150413efdfcc",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length:  0

The name of the AOR matters, as in the REGISTER it states “register to this AOR”. Right now it is registering to “6004” but you have it named “6004_aor”.

I changed the pjsip code like bellow

[6004]
type=endpoint
transport=transport-wss
context=from_external
disallow=all
allow=opus
allow=ulaw
aor=6004
auth=6004
media_encryption=dtls
dtls_auto_generate_cert=yes
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
direct_media=no
force_rport=yes
rewrite_contact=yes
webrtc=yes


[6004]
type=auth
auth_type=userpass
username=6004
password=6004

[6004]
type=aor
max_contacts1005
remove_existing=yes
contact=sip:6004@172.20.10.100

[provider_6004]
type=registration
outbound_auth=provider_6004_auth
server_uri=sip:172.16.1.1
client_uri=sip:6004@172.16.1.1
contact_user=6004
retry_interval=60
max_retries=10
transport=transport-wss


[provider_6004_auth]
type=auth
auth_type=userpass
username=6004
password=6004


Now I got bellow Notice and also not registered sip endpoint. What is the issue now??

[May 27 09:57:19] NOTICE[23047]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '"6004" <sip:6004@172.20.10.100>' failed for '172.20.8.117:10663' (callid: iphg48gimaihdg7e4qil) - No matching endpoint found
[May 27 09:57:19] NOTICE[23047]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '"6004" <sip:6004@172.20.10.100>' failed for '172.20.8.117:10663' (callid: iphg48gimaihdg7e4qil) - No matching endpoint found
[May 27 09:57:19] NOTICE[23047]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '"6004" <sip:6004@172.20.10.100>' failed for '172.20.8.117:10663' (callid: iphg48gimaihdg7e4qil) - Failed to authenticate

alos I received this error.


[May 27 12:25:04] WARNING[2588]: res_pjsip_outbound_registration.c:1065 schedule_retry: No response received from 'sip:172.16.1.1' on registration attempt to 'sip:6004@172.16.1.1', retrying in '60'

Now I successfully registered my sip account. When I am going to make outbound call, bellow error occurred. How fix it

Connected to Asterisk 20.5.0 currently running on dms (pid = 3252)
    -- Executing [0787733966@from_external:1] NoOp("PJSIP/6006-00000000", "Outbound call from "6006" <6006>") in new stack
    -- Executing [0787733966@from_external:2] Dial("PJSIP/6006-00000000", "PJSIP/0787733966@my_sip_trunk") in new stack
    -- Called PJSIP/0787733966@my_sip_trunk
[May 27 12:54:31] WARNING[3284]: res_pjsip_outbound_registration.c:1065 schedule_retry: No response received from 'sip:172.16.1.1' on registration attempt to 'sip:6006@172.16.1.1', retrying in '60'
[May 27 12:54:39] WARNING[3284]: res_pjsip_outbound_registration.c:1065 schedule_retry: No response received from 'sip:172.16.1.1' on registration attempt to 'sip:6005@172.16.1.1', retrying in '60'
[May 27 12:54:39] WARNING[3284]: res_pjsip_outbound_registration.c:1065 schedule_retry: No response received from 'sip:172.16.1.1' on registration attempt to 'sip:6004@172.16.1.1', retrying in '60'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0787733966@from_external:3] Hangup("PJSIP/6006-00000000", "") in new stack
  == Spawn extension (from_external, 0787733966, 3) exited non-zero on 'PJSIP/6006-00000000'
dms*CLI>

And this is log details

dms*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (2068 bytes) from WSS:172.20.8.117:5370 --->
INVITE sip:0787733966@172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.10;branch=z9hG4bK6349111
To: <sip:0787733966@172.20.10.100>
From: "6006" <sip:6006@172.20.10.100>;tag=35oeoac2jp
CSeq: 1 INVITE
Call-ID: glqg4c2a5f5ssqen931n
Max-Forwards: 70
Contact: <sip:6jgp8iii@192.0.2.10;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Type: application/sdp
Content-Length: 1492

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 896369571683998412 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 D5:E4:19:45:75:9D:42:46:88:42:9C:ED:8D:EA:C7:44:C5:6C:AF:1B:6B:AD:85:6F:EB:D9:96:0D:29:42:D4:0E
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 58087 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 124.43.67.53
a=candidate:0 1 UDP 2122252543 172.20.8.117 54029 typ host
a=candidate:2 1 TCP 2105524479 172.20.8.117 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 172.20.8.117 54030 typ host
a=candidate:2 2 TCP 2105524478 172.20.8.117 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 124.43.67.53 58087 typ srflx raddr 172.20.8.117 rport 54029
a=candidate:1 2 UDP 1686052862 124.43.67.53 54030 typ srflx raddr 172.20.8.117 rport 54030
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:71509168200dd99ca94a9b09cf0b61b7
a=ice-ufrag:fcf54dd2
a=mid:0
a=msid:{dbe401c1-fc64-47ca-88b0-07f931489a52} {707b8bb6-1c1d-456d-a17d-d24fa3e9944a}
a=rtcp:54030 IN IP4 124.43.67.53
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:315898643 cname:{d44df69f-efbc-4d08-a600-f9a8e5c052fc}

<--- Transmitting SIP response (463 bytes) to WSS:172.20.8.117:5370 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.10;rport=5370;received=172.20.8.117;branch=z9hG4bK6349111
Call-ID: glqg4c2a5f5ssqen931n
From: "6006" <sip:6006@172.20.10.100>;tag=35oeoac2jp
To: <sip:0787733966@172.20.10.100>;tag=z9hG4bK6349111
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1716794885/21f03345042267bbf6917376be703cd3",opaque="754d1e3f5c5e2339",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (285 bytes) from WSS:172.20.8.117:5370 --->
ACK sip:0787733966@172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.10;branch=z9hG4bK6349111
To: <sip:0787733966@172.20.10.100>;tag=z9hG4bK6349111
From: "6006" <sip:6006@172.20.10.100>;tag=35oeoac2jp
Call-ID: glqg4c2a5f5ssqen931n
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


<--- Received SIP request (2347 bytes) from WSS:172.20.8.117:5370 --->
INVITE sip:0787733966@172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.10;branch=z9hG4bK1587106
To: <sip:0787733966@172.20.10.100>
From: "6006" <sip:6006@172.20.10.100>;tag=35oeoac2jp
CSeq: 2 INVITE
Call-ID: glqg4c2a5f5ssqen931n
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="6006", realm="asterisk", nonce="1716794885/21f03345042267bbf6917376be703cd3", uri="sip:0787733966@172.20.10.100", response="46e28a8707435cf70039c50ffff10df0", opaque="754d1e3f5c5e2339", qop=auth, cnonce="rouk0ifh6ajt", nc=00000001
Contact: <sip:6jgp8iii@192.0.2.10;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Type: application/sdp
Content-Length: 1492

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 896369571683998412 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 D5:E4:19:45:75:9D:42:46:88:42:9C:ED:8D:EA:C7:44:C5:6C:AF:1B:6B:AD:85:6F:EB:D9:96:0D:29:42:D4:0E
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 58087 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 124.43.67.53
a=candidate:0 1 UDP 2122252543 172.20.8.117 54029 typ host
a=candidate:2 1 TCP 2105524479 172.20.8.117 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 172.20.8.117 54030 typ host
a=candidate:2 2 TCP 2105524478 172.20.8.117 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 124.43.67.53 58087 typ srflx raddr 172.20.8.117 rport 54029
a=candidate:1 2 UDP 1686052862 124.43.67.53 54030 typ srflx raddr 172.20.8.117 rport 54030
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:71509168200dd99ca94a9b09cf0b61b7
a=ice-ufrag:fcf54dd2
a=mid:0
a=msid:{dbe401c1-fc64-47ca-88b0-07f931489a52} {707b8bb6-1c1d-456d-a17d-d24fa3e9944a}
a=rtcp:54030 IN IP4 124.43.67.53
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:315898643 cname:{d44df69f-efbc-4d08-a600-f9a8e5c052fc}

<--- Transmitting SIP response (292 bytes) to WSS:172.20.8.117:5370 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.10;rport=5370;received=172.20.8.117;branch=z9hG4bK1587106
Call-ID: glqg4c2a5f5ssqen931n
From: "6006" <sip:6006@172.20.10.100>;tag=35oeoac2jp
To: <sip:0787733966@172.20.10.100>
CSeq: 2 INVITE
Server: Asterisk PBX 20.5.0
Content-Length:  0


    -- Executing [0787733966@from_external:1] NoOp("PJSIP/6006-00000002", "Outbound call from "6006" <6006>") in new stack
    -- Executing [0787733966@from_external:2] Dial("PJSIP/6006-00000002", "PJSIP/0787733966@my_sip_trunk") in new stack
    -- Called PJSIP/0787733966@my_sip_trunk
<--- Transmitting SIP request (972 bytes) to UDP:172.16.1.1:5060 --->
INVITE sip:0787733966@172.16.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjd8e6e352-9c35-4119-9dde-726012619c5c
From: "6006" <sip:6006@172.20.10.100>;tag=f4c6bbea-c2b6-4d3b-84bc-6f404833cc44
To: <sip:0787733966@172.16.1.1>
Contact: <sip:asterisk@172.20.10.100:5060>
Call-ID: 31a7cd2d-ade4-4af3-9c12-d4276970f85c
CSeq: 30123 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   294

v=0
o=- 619047279 619047279 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 12676 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP request (972 bytes) to UDP:172.16.1.1:5060 --->
INVITE sip:0787733966@172.16.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjd8e6e352-9c35-4119-9dde-726012619c5c
From: "6006" <sip:6006@172.20.10.100>;tag=f4c6bbea-c2b6-4d3b-84bc-6f404833cc44
To: <sip:0787733966@172.16.1.1>
Contact: <sip:asterisk@172.20.10.100:5060>
Call-ID: 31a7cd2d-ade4-4af3-9c12-d4276970f85c
CSeq: 30123 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   294

v=0
o=- 619047279 619047279 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 12676 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP request (972 bytes) to UDP:172.16.1.1:5060 --->
INVITE sip:0787733966@172.16.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjd8e6e352-9c35-4119-9dde-726012619c5c
From: "6006" <sip:6006@172.20.10.100>;tag=f4c6bbea-c2b6-4d3b-84bc-6f404833cc44
To: <sip:0787733966@172.16.1.1>
Contact: <sip:asterisk@172.20.10.100:5060>
Call-ID: 31a7cd2d-ade4-4af3-9c12-d4276970f85c
CSeq: 30123 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   294

v=0
o=- 619047279 619047279 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 12676 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP request (972 bytes) to UDP:172.16.1.1:5060 --->
INVITE sip:0787733966@172.16.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjd8e6e352-9c35-4119-9dde-726012619c5c
From: "6006" <sip:6006@172.20.10.100>;tag=f4c6bbea-c2b6-4d3b-84bc-6f404833cc44
To: <sip:0787733966@172.16.1.1>
Contact: <sip:asterisk@172.20.10.100:5060>
Call-ID: 31a7cd2d-ade4-4af3-9c12-d4276970f85c
CSeq: 30123 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   294

v=0
o=- 619047279 619047279 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 12676 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP request (972 bytes) to UDP:172.16.1.1:5060 --->
INVITE sip:0787733966@172.16.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjd8e6e352-9c35-4119-9dde-726012619c5c
From: "6006" <sip:6006@172.20.10.100>;tag=f4c6bbea-c2b6-4d3b-84bc-6f404833cc44
To: <sip:0787733966@172.16.1.1>
Contact: <sip:asterisk@172.20.10.100:5060>
Call-ID: 31a7cd2d-ade4-4af3-9c12-d4276970f85c
CSeq: 30123 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   294

v=0
o=- 619047279 619047279 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 12676 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP request (972 bytes) to UDP:172.16.1.1:5060 --->
INVITE sip:0787733966@172.16.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjd8e6e352-9c35-4119-9dde-726012619c5c
From: "6006" <sip:6006@172.20.10.100>;tag=f4c6bbea-c2b6-4d3b-84bc-6f404833cc44
To: <sip:0787733966@172.16.1.1>
Contact: <sip:asterisk@172.20.10.100:5060>
Call-ID: 31a7cd2d-ade4-4af3-9c12-d4276970f85c
CSeq: 30123 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   294

v=0
o=- 619047279 619047279 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 12676 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP request (534 bytes) to UDP:172.16.1.1:5060 --->
REGISTER sip:172.16.1.1 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjcc899549-6a9a-4de3-8bf7-02d7ae77a901
From: <sip:6006@172.16.1.1>;tag=5319f62e-f0e6-4c35-a611-138366220a9f
To: <sip:6006@172.16.1.1>
Call-ID: dadbc277-1b0a-4c80-a1a7-9d6ecdbcfdf7
CSeq: 29847 REGISTER
Contact: <sip:6006@172.20.10.100:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (534 bytes) to UDP:172.16.1.1:5060 --->
REGISTER sip:172.16.1.1 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjcc899549-6a9a-4de3-8bf7-02d7ae77a901
From: <sip:6006@172.16.1.1>;tag=5319f62e-f0e6-4c35-a611-138366220a9f
To: <sip:6006@172.16.1.1>
Call-ID: dadbc277-1b0a-4c80-a1a7-9d6ecdbcfdf7
CSeq: 29847 REGISTER
Contact: <sip:6006@172.20.10.100:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (534 bytes) to UDP:172.16.1.1:5060 --->
REGISTER sip:172.16.1.1 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjcc899549-6a9a-4de3-8bf7-02d7ae77a901
From: <sip:6006@172.16.1.1>;tag=5319f62e-f0e6-4c35-a611-138366220a9f
To: <sip:6006@172.16.1.1>
Call-ID: dadbc277-1b0a-4c80-a1a7-9d6ecdbcfdf7
CSeq: 29847 REGISTER
Contact: <sip:6006@172.20.10.100:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (972 bytes) to UDP:172.16.1.1:5060 --->
INVITE sip:0787733966@172.16.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPjd8e6e352-9c35-4119-9dde-726012619c5c
From: "6006" <sip:6006@172.20.10.100>;tag=f4c6bbea-c2b6-4d3b-84bc-6f404833cc44
To: <sip:0787733966@172.16.1.1>
Contact: <sip:asterisk@172.20.10.100:5060>
Call-ID: 31a7cd2d-ade4-4af3-9c12-d4276970f85c
CSeq: 30123 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   294

v=0
o=- 619047279 619047279 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 12676 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0787733966@from_external:3] Hangup("PJSIP/6006-00000002", "") in new stack
  == Spawn extension (from_external, 0787733966, 3) exited non-zero on 'PJSIP/6006-00000002'
<--- Transmitting SIP response (366 bytes) to WSS:172.20.8.117:5370 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/WSS 192.0.2.10;rport=5370;received=172.20.8.117;branch=z9hG4bK1587106
Call-ID: glqg4c2a5f5ssqen931n
From: "6006" <sip:6006@172.20.10.100>;tag=35oeoac2jp
To: <sip:0787733966@172.20.10.100>;tag=d5de3407-8c6f-414c-bee0-3fec964e4c0b
CSeq: 2 INVITE
Server: Asterisk PBX 20.5.0
Reason: Q.850;cause=18
Content-Length:  0


<--- Received SIP request (307 bytes) from WSS:172.20.8.117:5370 --->
ACK sip:0787733966@172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.10;branch=z9hG4bK1587106
To: <sip:0787733966@172.20.10.100>;tag=d5de3407-8c6f-414c-bee0-3fec964e4c0b
From: "6006" <sip:6006@172.20.10.100>;tag=35oeoac2jp
Call-ID: glqg4c2a5f5ssqen931n
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0



Asterisk is sending a SIP call to 172.16.1.1:5060 and getting no response.

I solved that problem. Now, My outbound is registered successfully. Bellow the pjsip code I was use for it.

[6006]
type=auth
auth_type=userpass
username=6006
password=6006

[6006]
type=registration
server_uri=sip:172.20.10.100
client_uri=sip:6006@172.20.10.86
outbound_auth=6006
contact_user=6006
retry_interval=60
max_retries=10
transport=transport-wss

[6006]
type=endpoint
context=outbound-calls
disallow=all
allow=opus
allow=ulaw
auth=6006
aors=6006
dtls_auto_generate_cert=yes
webrtc=yes
;rtp_symmetric=yes
;force_rport=yes
;rewrite_contact=yes

[6006]
type=aor
max_contacts=5
remove_existing=yes
contact=sip:172.20.10.86


;=================================================================================================

[my_sip_trunk]
type=endpoint
context=outbound-calls
disallow=all
allow=ulaw
allow=opus
aors=my_sip_trunk

[my_sip_trunk]
type=aor
contact=sip:172.20.10.86

[my_sip_trunk]
type=identify
endpoint=my_sip_trunk
;match=172.20.10.100
;match=172.16.1.1
match=172.20.10.86

But problem is, When I dialed client number (07877xxxxx) using browser phone, It started calling. But not received call to destination number (07877xxxxxx). In asterisk CLI show bellow logs.

 Executing [0787733966@outbound-calls:1] NoOp("PJSIP/6006-00000000", "Making outbound call to 0787733966") in new stack
    -- Executing [0787733966@outbound-calls:2] Dial("PJSIP/6006-00000000", "PJSIP/0787733966@my_sip_trunk,30,g") in new stack
    -- Called PJSIP/0787733966@my_sip_trunk
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0787733966@outbound-calls:3] BackGround("PJSIP/6006-00000000", "welcome_en") in new stack
       > 0x7ff54c075610 -- Strict RTP learning after remote address set to: 172.20.8.117:52965
       > 0x7ff54c075610 -- Strict RTP learning after ICE completion
       > 0x7ff54c075610 -- Strict RTP learning after remote address set to: 172.20.8.117:52965
    -- <PJSIP/6006-00000000> Playing 'welcome_en.slin48' (language 'en')
       > 0x7ff54c075610 -- Strict RTP switching to RTP target address 172.20.8.117:52965 as source
       > 0x7ff54c075610 -- Strict RTP learning complete - Locking on source address 172.20.8.117:52965
    -- Executing [0787733966@outbound-calls:4] Hangup("PJSIP/6006-00000000", "") in new stack
  == Spawn extension (outbound-calls, 0787733966, 4) exited non-zero on 'PJSIP/6006-00000000'

What is the issue?

Insufficient information. Your provider rejected the call, but you have not provided the pjsip set logger on type output to show the reason they gave.

It is also possible that they are not responding to the qualify checks.

This is the logger.

<--- Received SIP request (2071 bytes) from WSS:172.20.8.117:4923 --->
INVITE sip:0787733966@172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.168;branch=z9hG4bK135141
To: <sip:0787733966@172.20.10.100>
From: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
CSeq: 1 INVITE
Call-ID: jj0fjl2mfm12td9imfnf
Max-Forwards: 70
Contact: <sip:vtmce3a4@192.0.2.168;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Type: application/sdp
Content-Length: 1494

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 1065138139821588660 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 A6:0F:4A:A7:EB:DE:1C:60:AB:05:7C:7E:3B:45:CB:B5:3F:65:3F:F6:DB:46:27:2E:2C:CF:7A:9F:A9:B3:64:44
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 54636 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 124.43.67.53
a=candidate:0 1 UDP 2122252543 172.20.8.117 54636 typ host
a=candidate:2 1 TCP 2105524479 172.20.8.117 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 172.20.8.117 54637 typ host
a=candidate:2 2 TCP 2105524478 172.20.8.117 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 124.43.67.53 54636 typ srflx raddr 172.20.8.117 rport 54636
a=candidate:1 2 UDP 1686052862 124.43.67.53 33475 typ srflx raddr 172.20.8.117 rport 54637
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:6373ba0d577f4094048eb2ed2158f8d2
a=ice-ufrag:6310e3cd
a=mid:0
a=msid:{01b208f4-5889-4206-85c7-e65b545dfd77} {4e892966-8dbb-4aa4-81d6-35090370a72c}
a=rtcp:33475 IN IP4 124.43.67.53
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:4200227931 cname:{53f3f8b7-1334-4295-9098-81d6280c037e}

<--- Transmitting SIP response (462 bytes) to WSS:172.20.8.117:4923 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.168;rport=4923;received=172.20.8.117;branch=z9hG4bK135141
Call-ID: jj0fjl2mfm12td9imfnf
From: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
To: <sip:0787733966@172.20.10.100>;tag=z9hG4bK135141
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1716881218/a01f2a66da8bed41962a0a191f5f1b41",opaque="5d3be8007ab50415",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (284 bytes) from WSS:172.20.8.117:4923 --->
ACK sip:0787733966@172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.168;branch=z9hG4bK135141
To: <sip:0787733966@172.20.10.100>;tag=z9hG4bK135141
From: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
Call-ID: jj0fjl2mfm12td9imfnf
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


<--- Received SIP request (2351 bytes) from WSS:172.20.8.117:4923 --->
INVITE sip:0787733966@172.20.10.100 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.168;branch=z9hG4bK4062354
To: <sip:0787733966@172.20.10.100>
From: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
CSeq: 2 INVITE
Call-ID: jj0fjl2mfm12td9imfnf
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="6006", realm="asterisk", nonce="1716881218/a01f2a66da8bed41962a0a191f5f1b41", uri="sip:0787733966@172.20.10.100", response="c3e40bc13bf150de7ae4652bb7535cc1", opaque="5d3be8007ab50415", qop=auth, cnonce="kom9uj6cvqh9", nc=00000001
Contact: <sip:vtmce3a4@192.0.2.168;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Type: application/sdp
Content-Length: 1494

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 1065138139821588660 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 A6:0F:4A:A7:EB:DE:1C:60:AB:05:7C:7E:3B:45:CB:B5:3F:65:3F:F6:DB:46:27:2E:2C:CF:7A:9F:A9:B3:64:44
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 54636 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 124.43.67.53
a=candidate:0 1 UDP 2122252543 172.20.8.117 54636 typ host
a=candidate:2 1 TCP 2105524479 172.20.8.117 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 172.20.8.117 54637 typ host
a=candidate:2 2 TCP 2105524478 172.20.8.117 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 124.43.67.53 54636 typ srflx raddr 172.20.8.117 rport 54636
a=candidate:1 2 UDP 1686052862 124.43.67.53 33475 typ srflx raddr 172.20.8.117 rport 54637
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:6373ba0d577f4094048eb2ed2158f8d2
a=ice-ufrag:6310e3cd
a=mid:0
a=msid:{01b208f4-5889-4206-85c7-e65b545dfd77} {4e892966-8dbb-4aa4-81d6-35090370a72c}
a=rtcp:33475 IN IP4 124.43.67.53
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:4200227931 cname:{53f3f8b7-1334-4295-9098-81d6280c037e}

<--- Transmitting SIP response (293 bytes) to WSS:172.20.8.117:4923 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.168;rport=4923;received=172.20.8.117;branch=z9hG4bK4062354
Call-ID: jj0fjl2mfm12td9imfnf
From: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
To: <sip:0787733966@172.20.10.100>
CSeq: 2 INVITE
Server: Asterisk PBX 20.5.0
Content-Length:  0


    -- Executing [0787733966@outbound-calls:1] NoOp("PJSIP/6006-00000002", "Making outbound call to 0787733966") in new stack
    -- Executing [0787733966@outbound-calls:2] Dial("PJSIP/6006-00000002", "PJSIP/0787733966@my_sip_trunk,30,g") in new stack
    -- Called PJSIP/0787733966@my_sip_trunk
<--- Transmitting SIP request (973 bytes) to UDP:172.20.10.86:5060 --->
INVITE sip:0787733966@172.20.10.86 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPj23d67fe6-aa15-4ec8-bed5-bee03d51fbeb
From: "6006" <sip:6006@172.20.10.100>;tag=caee6afc-8c6e-4b74-98bd-c1c3ea755d59
To: <sip:0787733966@172.20.10.86>
Contact: <sip:asterisk@172.20.10.100:5060>
Call-ID: b2a14992-4948-4b47-83a0-7ff1d240fe3e
CSeq: 13515 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   296

v=0
o=- 1838055131 1838055131 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 13584 RTP/AVP 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP response (558 bytes) from UDP:172.20.10.86:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPj23d67fe6-aa15-4ec8-bed5-bee03d51fbeb
From: "6006" <sip:6006@172.20.10.100>;tag=caee6afc-8c6e-4b74-98bd-c1c3ea755d59
To: <sip:0787733966@172.20.10.86>;tag=1c387007413
Call-ID: b2a14992-4948-4b47-83a0-7ff1d240fe3e
CSeq: 13515 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.40A.029.008
Content-Length: 0


<--- Received SIP response (614 bytes) from UDP:172.20.10.86:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPj23d67fe6-aa15-4ec8-bed5-bee03d51fbeb
From: "6006" <sip:6006@172.20.10.100>;tag=caee6afc-8c6e-4b74-98bd-c1c3ea755d59
To: <sip:0787733966@172.20.10.86>;tag=1c387007413
Call-ID: b2a14992-4948-4b47-83a0-7ff1d240fe3e
CSeq: 13515 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.40A.029.008
Reason: Q.850 ;cause=21 ;text="local"
Content-Length: 0


<--- Transmitting SIP request (405 bytes) to UDP:172.20.10.86:5060 --->
ACK sip:0787733966@172.20.10.86 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPj23d67fe6-aa15-4ec8-bed5-bee03d51fbeb
From: "6006" <sip:6006@172.20.10.100>;tag=caee6afc-8c6e-4b74-98bd-c1c3ea755d59
To: <sip:0787733966@172.20.10.86>;tag=1c387007413
Call-ID: b2a14992-4948-4b47-83a0-7ff1d240fe3e
CSeq: 13515 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0787733966@outbound-calls:3] BackGround("PJSIP/6006-00000002", "welcome_en") in new stack
       > 0x7ff52000ef30 -- Strict RTP learning after remote address set to: 124.43.67.53:54636
<--- Transmitting SIP response (1492 bytes) to WSS:172.20.8.117:4923 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.168;rport=4923;received=172.20.8.117;branch=z9hG4bK4062354
Call-ID: jj0fjl2mfm12td9imfnf
From: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
To: <sip:0787733966@172.20.10.100>;tag=92a57f85-4613-4930-b221-8a5a50edc3ae
CSeq: 2 INVITE
Server: Asterisk PBX 20.5.0
Contact: <sip:172.20.10.100:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   921

v=0
o=- 2898870452 2 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 13814 UDP/TLS/RTP/SAVPF 109 0 101
a=connection:new
a=setup:active
a=fingerprint:SHA-256 B4:CD:7B:81:1E:8E:95:31:13:5D:66:9A:B3:99:7A:ED:68:DF:BD:C6:01:9A:25:39:B9:58:95:05:54:07:D6:A4
a=ice-ufrag:700fce993c81e051246a955f53c50755
a=ice-pwd:21b23c2e2d5566747225ebe006043e6f
a=candidate:Hac140a64 1 UDP 2130706431 172.20.10.100 13814 typ host
a=candidate:H8259629a 1 UDP 2130706431 fe80::250:56ff:feb0:1750 13814 typ host
a=rtpmap:109 opus/48000/2
a=fmtp:109 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1371411112 cname:234dc70f-b4d9-487b-bf85-18d577abb7fc
a=msid:ebd3a363-2150-4b93-8224-8e10d73dc39b ad0d8325-11be-4663-bde5-e8ca3028aa71
a=rtcp-fb:* transport-cc
a=mid:0

       > 0x7ff52000ef30 -- Strict RTP learning after ICE completion
<--- Received SIP request (468 bytes) from WSS:172.20.8.117:4923 --->
ACK sip:172.20.10.100:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.168;branch=z9hG4bK7615982
To: <sip:0787733966@172.20.10.100>;tag=92a57f85-4613-4930-b221-8a5a50edc3ae
From: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
CSeq: 2 ACK
Call-ID: jj0fjl2mfm12td9imfnf
Max-Forwards: 70
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Length: 0


       > 0x7ff52000ef30 -- Strict RTP learning after remote address set to: 172.20.8.117:54636
    -- <PJSIP/6006-00000002> Playing 'welcome_en.slin48' (language 'en')
       > 0x7ff52000ef30 -- Strict RTP switching to RTP target address 172.20.8.117:54636 as source
       > 0x7ff52000ef30 -- Strict RTP learning complete - Locking on source address 172.20.8.117:54636
    -- Executing [0787733966@outbound-calls:4] Hangup("PJSIP/6006-00000002", "") in new stack
  == Spawn extension (outbound-calls, 0787733966, 4) exited non-zero on 'PJSIP/6006-00000002'
<--- Transmitting SIP request (438 bytes) to WSS:172.20.8.117:4923 --->
BYE sip:vtmce3a4@172.20.8.117:4923;transport=ws;ob SIP/2.0
Via: SIP/2.0/WSS 172.20.10.100:8089;rport;branch=z9hG4bKPjbc493876-94cf-4dd5-a1f5-e5fea211ef76;alias
From: <sip:0787733966@172.20.10.100>;tag=92a57f85-4613-4930-b221-8a5a50edc3ae
To: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
Call-ID: jj0fjl2mfm12td9imfnf
CSeq: 20023 BYE
Reason: Q.850;cause=21
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP response (471 bytes) from WSS:172.20.8.117:4923 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 172.20.10.100:8089;rport;branch=z9hG4bKPjbc493876-94cf-4dd5-a1f5-e5fea211ef76;alias
From: <sip:0787733966@172.20.10.100>;tag=92a57f85-4613-4930-b221-8a5a50edc3ae
To: "6006" <sip:6006@172.20.10.100>;tag=oamh0rigq7
CSeq: 20023 BYE
Call-ID: jj0fjl2mfm12td9imfnf
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Length: 0

You will need to ask Audiocodes what their product means by that, or look at the logs that it produces.

I solved the problem. After I add the from_user into endpoint of my_sip_trunck of pjsip, call was succeeded.

[my_sip_trunk]
type=endpoint
context=outbound-calls
disallow=all
allow=ulaw
allow=opus
aors=my_sip_trunk
from_user=11xxxxxxx

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