SIP 503 Response

Asterisk Asterisk SIP
1

I am also getting the same error:

Comment -------- After I pick up the call, I see the following logs.
<— Received SIP response (417 bytes) from WS:127.0.0.1:48224 —>
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPj13fc9b8c-6ae1-4474-97f8-be3b3fdd35a5;alias
From: “1002” sip:1002@ip-10-0-16-85;tag=a69ad0a4-737c-43b6-9e8d-54c1c453285b
To: sip:h6oeguhk@127.0.0.1;tag=evc3jod1uc
CSeq: 5780 INVITE
Call-ID: ce5f25f6-723f-4264-b946-a21cafc88f57
Supported: outbound
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<— Transmitting SIP request (415 bytes) to WS:127.0.0.1:48224 —>
ACK sip:h6oeguhk@127.0.0.1:48224;transport=WS SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPj13fc9b8c-6ae1-4474-97f8-be3b3fdd35a5;alias
From: “1002” sip:1002@ip-10-0-16-85;tag=a69ad0a4-737c-43b6-9e8d-54c1c453285b
To: sip:h6oeguhk@127.0.0.1;tag=evc3jod1uc
Call-ID: ce5f25f6-723f-4264-b946-a21cafc88f57
CSeq: 5780 ACK
Max-Forwards: 70
User-Agent: FPBX-16.0.26(18.15.0)
Content-Length: 0

<— Transmitting SIP response (533 bytes) to WS:127.0.0.1:55620 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/WSS 192.0.2.119;rport=55620;received=127.0.0.1;branch=z9hG4bK4556358
Call-ID: e12cqd3oq2kuaijqckct
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
To: sip:1001@nj.test.com;tag=c28d5129-ef12-4299-9e7c-e0c69897ed7c
CSeq: 2 INVITE
Server: FPBX-16.0.26(18.15.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Reason: Q.850;cause=34
P-Asserted-Identity: “1001” sip:1001@nj.test.com
Content-Length: 0

== Spawn extension (macro-exten-vm, s-NOANSWER, 3) exited non-zero on ‘PJSIP/1002-00000025’ in macro ‘exten-vm’
== Spawn extension (ext-local, 1001, 3) exited non-zero on ‘PJSIP/1002-00000025’
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘PJSIP/1002-00000025’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/1002-00000025’
<— Received SIP request (299 bytes) from WS:127.0.0.1:55620 —>
ACK sip:1001@nj.test.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.119;branch=z9hG4bK4556358
To: sip:1001@nj.test.com;tag=c28d5129-ef12-4299-9e7c-e0c69897ed7c
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
Call-ID: e12cqd3oq2kuaijqckct
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

2

You are getting nothing like the same error, and if you had been you should have used the solution from the original thread. In particular, in your case the called device rejected the call, but in the original one it accepted it.

For a definition of 480, see:

But you need to start your own topic, with a full description of your problem. I’ve asked for your posting to be moved to a new thread.

3

Here is the complete logs:

<— Transmitting SIP request (415 bytes) to UDP:46.19.209.14:5060 —>
OPTIONS sip:46.19.209.14 SIP/2.0
Via: SIP/2.0/UDP 54.87.215.201:5060;rport;branch=z9hG4bKPjd6c25b58-0b25-4fe9-915f-bf59480e0303
From: sip:BuyDID_2@10.0.16.85;tag=964177d7-267f-4817-9959-52d53b2fd4a5
To: sip:46.19.209.14
Contact: sip:BuyDID_2@54.87.215.201:5060
Call-ID: 7135c28e-c32c-463a-8f91-ea8c7dd45f28
CSeq: 55846 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.26(18.15.0)
Content-Length: 0

<— Received SIP response (411 bytes) from UDP:46.19.209.14:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.87.215.201:5060;rport=5060;branch=z9hG4bKPjd6c25b58-0b25-4fe9-915f-bf59480e0303;received=54.87.215.201
From: sip:BuyDID_2@10.0.16.85;tag=964177d7-267f-4817-9959-52d53b2fd4a5
To: sip:46.19.209.14;tag=815cbf4cfc3e75322dd43ade78dddabe.cd6b
Call-ID: 7135c28e-c32c-463a-8f91-ea8c7dd45f28
CSeq: 55846 OPTIONS
Server: RS OOD receiver node v625 shmele
Content-Length: 0

<— Received SIP request (2785 bytes) from WS:127.0.0.1:55620 —>
INVITE sip:1001@nj.test.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.119;branch=z9hG4bK4275910
To: sip:1001@nj.test.com
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
CSeq: 1 INVITE
Call-ID: e12cqd3oq2kuaijqckct
Max-Forwards: 70
Contact: sip:980ml152@192.0.2.119;transport=wss;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Type: application/sdp
Content-Length: 2298

v=0
o=- 5800509607678442233 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS f3a0866f-81a4-48d4-8f19-51bba6b3495f
m=audio 61580 UDP/TLS/RTP/SAVPF 111 63 103 9 0 8 105 13 110 113 126
c=IN IP4 76.226.93.98
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2063314161 1 udp 2122262783 2600:1700:20c1:1990:95d8:8a1b:ca14:e1b2 59307 typ host generation 0 network-id 2 network-cost 50
a=candidate:1681997092 1 udp 2122194687 192.168.1.66 61580 typ host generation 0 network-id 1 network-cost 50
a=candidate:2454799600 1 udp 1685987071 76.226.93.98 61580 typ srflx raddr 192.168.1.66 rport 61580 generation 0 network-id 1 network-cost 50
a=candidate:880300033 1 tcp 1518283007 2600:1700:20c1:1990:95d8:8a1b:ca14:e1b2 9 typ host tcptype active generation 0 network-id 2 network-cost 50
a=candidate:717406676 1 tcp 1518214911 192.168.1.66 9 typ host tcptype active generation 0 network-id 1 network-cost 50
a=ice-ufrag:SqKJ
a=ice-pwd:AdCYS9Wp61watv4G2vNDMPvV
a=ice-options:trickle
a=fingerprint:sha-256 B7:00:B8:75:D0:98:14:37:C4:05:F9:1D:47:BD:FB:FF:9F:7E:4A:C9:7B:48:F6:C1:3D:EB:26:F4:B5:7A:E8:64
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:f3a0866f-81a4-48d4-8f19-51bba6b3495f 64dacaa8-87a4-4ed0-ab6e-1a4034ececde
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3927429298 cname:5uK9o/AroNq5WBir
a=ssrc:3927429298 msid:f3a0866f-81a4-48d4-8f19-51bba6b3495f 64dacaa8-87a4-4ed0-ab6e-1a4034ececde
a=ssrc:3927429298 mslabel:f3a0866f-81a4-48d4-8f19-51bba6b3495f
a=ssrc:3927429298 label:64dacaa8-87a4-4ed0-ab6e-1a4034ececde

<— Transmitting SIP response (461 bytes) to WS:127.0.0.1:55620 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.119;rport=55620;received=127.0.0.1;branch=z9hG4bK4275910
Call-ID: e12cqd3oq2kuaijqckct
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
To: sip:1001@nj.test.com;tag=z9hG4bK4275910
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1669491167/45c6c0ad7dddbd128c8bd7bac18d0957”,opaque=“468875bb6a14fa38”,algorithm=MD5,qop=“auth”
Server: FPBX-16.0.26(18.15.0)
Content-Length: 0

<— Received SIP request (277 bytes) from WS:127.0.0.1:55620 —>
ACK sip:1001@nj.test.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.119;branch=z9hG4bK4275910
To: sip:1001@nj.test.com;tag=z9hG4bK4275910
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
Call-ID: e12cqd3oq2kuaijqckct
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<— Received SIP request (3058 bytes) from WS:127.0.0.1:55620 —>
INVITE sip:1001@nj.test.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.119;branch=z9hG4bK4556358
To: sip:1001@nj.test.com
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
CSeq: 2 INVITE
Call-ID: e12cqd3oq2kuaijqckct
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“1002”, realm=“asterisk”, nonce=“1669491167/45c6c0ad7dddbd128c8bd7bac18d0957”, uri="sip:1001@nj.test.com", response=“608138080e495a204416d7d831c805fb”, opaque=“468875bb6a14fa38”, qop=auth, cnonce=“dpo2a6ikhp6d”, nc=00000001
Contact: sip:980ml152@192.0.2.119;transport=wss;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Type: application/sdp
Content-Length: 2298

v=0
o=- 5800509607678442233 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS f3a0866f-81a4-48d4-8f19-51bba6b3495f
m=audio 61580 UDP/TLS/RTP/SAVPF 111 63 103 9 0 8 105 13 110 113 126
c=IN IP4 76.226.93.98
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2063314161 1 udp 2122262783 2600:1700:20c1:1990:95d8:8a1b:ca14:e1b2 59307 typ host generation 0 network-id 2 network-cost 50
a=candidate:1681997092 1 udp 2122194687 192.168.1.66 61580 typ host generation 0 network-id 1 network-cost 50
a=candidate:2454799600 1 udp 1685987071 76.226.93.98 61580 typ srflx raddr 192.168.1.66 rport 61580 generation 0 network-id 1 network-cost 50
a=candidate:880300033 1 tcp 1518283007 2600:1700:20c1:1990:95d8:8a1b:ca14:e1b2 9 typ host tcptype active generation 0 network-id 2 network-cost 50
a=candidate:717406676 1 tcp 1518214911 192.168.1.66 9 typ host tcptype active generation 0 network-id 1 network-cost 50
a=ice-ufrag:SqKJ
a=ice-pwd:AdCYS9Wp61watv4G2vNDMPvV
a=ice-options:trickle
a=fingerprint:sha-256 B7:00:B8:75:D0:98:14:37:C4:05:F9:1D:47:BD:FB:FF:9F:7E:4A:C9:7B:48:F6:C1:3D:EB:26:F4:B5:7A:E8:64
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:f3a0866f-81a4-48d4-8f19-51bba6b3495f 64dacaa8-87a4-4ed0-ab6e-1a4034ececde
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3927429298 cname:5uK9o/AroNq5WBir
a=ssrc:3927429298 msid:f3a0866f-81a4-48d4-8f19-51bba6b3495f 64dacaa8-87a4-4ed0-ab6e-1a4034ececde
a=ssrc:3927429298 mslabel:f3a0866f-81a4-48d4-8f19-51bba6b3495f
a=ssrc:3927429298 label:64dacaa8-87a4-4ed0-ab6e-1a4034ececde

<— Transmitting SIP response (290 bytes) to WS:127.0.0.1:55620 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.119;rport=55620;received=127.0.0.1;branch=z9hG4bK4556358
Call-ID: e12cqd3oq2kuaijqckct
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
To: sip:1001@nj.test.com
CSeq: 2 INVITE
Server: FPBX-16.0.26(18.15.0)
Content-Length: 0

== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
== Spawn extension (from-internal, 1001, 1) exited non-zero on ‘PJSIP/1001-00000026’
<— Transmitting SIP response (541 bytes) to WS:127.0.0.1:55620 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 192.0.2.119;rport=55620;received=127.0.0.1;branch=z9hG4bK4556358
Call-ID: e12cqd3oq2kuaijqckct
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
To: sip:1001@nj.test.com;tag=c28d5129-ef12-4299-9e7c-e0c69897ed7c
CSeq: 2 INVITE
Server: FPBX-16.0.26(18.15.0)
Contact: sip:127.0.0.1:8080;transport=ws
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
P-Asserted-Identity: “1001” sip:1001@nj.test.com
Content-Length: 0

== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
<— Transmitting SIP request (1542 bytes) to WS:127.0.0.1:48224 —>
INVITE sip:h6oeguhk@127.0.0.1:48224;transport=WS SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPj13fc9b8c-6ae1-4474-97f8-be3b3fdd35a5;alias
From: “1002” sip:1002@ip-10-0-16-85;tag=a69ad0a4-737c-43b6-9e8d-54c1c453285b
To: sip:h6oeguhk@127.0.0.1
Contact: sip:asterisk@ip-10-0-16-85:5060;transport=ws
Call-ID: ce5f25f6-723f-4264-b946-a21cafc88f57
CSeq: 5780 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: “1002” sip:1002@ip-10-0-16-85
Max-Forwards: 70
User-Agent: FPBX-16.0.26(18.15.0)
Content-Type: application/sdp
Content-Length: 787

v=0
o=- 995678637 995678637 IN IP4 10.0.16.85
s=Asterisk
c=IN IP4 10.0.16.85
t=0 0
a=group:BUNDLE audio-0
m=audio 25306 RTP/AVPF 0 8 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:j0qKwt7oCyJQsg8CcMj9x+msJSrkqXCW8WCcNj2y
a=ice-ufrag:4dd5624329b7048813335f6d11b1a6d4
a=ice-pwd:2b23ae512af86ed26c6d9b48569acfe2
a=candidate:Ha001055 1 UDP 2130706431 10.0.16.85 25306 typ host
a=candidate:H8eb495ed 1 UDP 2130706431 fe80::43a:42ff:feaa:5751 25306 typ host
a=candidate:S3657d7c9 1 UDP 1694498815 54.87.215.201 25306 typ srflx raddr 10.0.16.85 rport 25306
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:1401290800 cname:d908f293-bc67-4f5a-a7bb-815d2282cecf
a=mid:audio-0

<— Received SIP response (385 bytes) from WS:127.0.0.1:48224 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPj13fc9b8c-6ae1-4474-97f8-be3b3fdd35a5;alias
From: “1002” sip:1002@ip-10-0-16-85;tag=a69ad0a4-737c-43b6-9e8d-54c1c453285b
To: sip:h6oeguhk@127.0.0.1
CSeq: 5780 INVITE
Call-ID: ce5f25f6-723f-4264-b946-a21cafc88f57
Supported: outbound
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<— Received SIP response (452 bytes) from WS:127.0.0.1:48224 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPj13fc9b8c-6ae1-4474-97f8-be3b3fdd35a5;alias
From: “1002” sip:1002@ip-10-0-16-85;tag=a69ad0a4-737c-43b6-9e8d-54c1c453285b
To: sip:h6oeguhk@127.0.0.1;tag=evc3jod1uc
CSeq: 5780 INVITE
Call-ID: ce5f25f6-723f-4264-b946-a21cafc88f57
Supported: outbound
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Contact: sip:h6oeguhk@192.0.2.247;transport=wss
Content-Length: 0

<— Transmitting SIP response (541 bytes) to WS:127.0.0.1:55620 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 192.0.2.119;rport=55620;received=127.0.0.1;branch=z9hG4bK4556358
Call-ID: e12cqd3oq2kuaijqckct
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
To: sip:1001@nj.test.com;tag=c28d5129-ef12-4299-9e7c-e0c69897ed7c
CSeq: 2 INVITE
Server: FPBX-16.0.26(18.15.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: sip:127.0.0.1:8080;transport=ws
P-Asserted-Identity: “1001” sip:1001@nj.test.com
Content-Length: 0

ip-10-0-16-85CLI>
ip-10-0-16-85CLI>
ip-10-0-16-85CLI>
ip-10-0-16-85CLI>
ip-10-0-16-85CLI>
ip-10-0-16-85CLI>
ip-10-0-16-85*CLI>
<— Transmitting SIP request (415 bytes) to UDP:46.19.210.14:5060 —>
OPTIONS sip:46.19.210.14 SIP/2.0
Via: SIP/2.0/UDP 54.87.215.201:5060;rport;branch=z9hG4bKPj020ee283-e4bd-4e10-82e8-80813ac899a6
From: sip:BuyDID_1@10.0.16.85;tag=8e8b5919-1aa7-4bab-bf9a-4b738a049df4
To: sip:46.19.210.14
Contact: sip:BuyDID_1@54.87.215.201:5060
Call-ID: 25c8c16b-1624-432e-a89d-66b6d0dfcef2
CSeq: 47833 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.26(18.15.0)
Content-Length: 0

ip-10-0-16-85*CLI>
<— Received SIP response (411 bytes) from UDP:46.19.210.14:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.87.215.201:5060;rport=5060;branch=z9hG4bKPj020ee283-e4bd-4e10-82e8-80813ac899a6;received=54.87.215.201
From: sip:BuyDID_1@10.0.16.85;tag=8e8b5919-1aa7-4bab-bf9a-4b738a049df4
To: sip:46.19.210.14;tag=96ef325e4e1162bc586f6ca7aebc6156.5870
Call-ID: 25c8c16b-1624-432e-a89d-66b6d0dfcef2
CSeq: 47833 OPTIONS
Server: RS OOD receiver node v625 shmele
Content-Length: 0

After accept the call:

Comment -------- After I pick up the call, I see the following logs.
<— Received SIP response (417 bytes) from WS:127.0.0.1:48224 —>
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPj13fc9b8c-6ae1-4474-97f8-be3b3fdd35a5;alias
From: “1002” sip:1002@ip-10-0-16-85;tag=a69ad0a4-737c-43b6-9e8d-54c1c453285b
To: sip:h6oeguhk@127.0.0.1;tag=evc3jod1uc
CSeq: 5780 INVITE
Call-ID: ce5f25f6-723f-4264-b946-a21cafc88f57
Supported: outbound
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<— Transmitting SIP request (415 bytes) to WS:127.0.0.1:48224 —>
ACK sip:h6oeguhk@127.0.0.1:48224;transport=WS SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPj13fc9b8c-6ae1-4474-97f8-be3b3fdd35a5;alias
From: “1002” sip:1002@ip-10-0-16-85;tag=a69ad0a4-737c-43b6-9e8d-54c1c453285b
To: sip:h6oeguhk@127.0.0.1;tag=evc3jod1uc
Call-ID: ce5f25f6-723f-4264-b946-a21cafc88f57
CSeq: 5780 ACK
Max-Forwards: 70
User-Agent: FPBX-16.0.26(18.15.0)
Content-Length: 0

<— Transmitting SIP response (533 bytes) to WS:127.0.0.1:55620 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/WSS 192.0.2.119;rport=55620;received=127.0.0.1;branch=z9hG4bK4556358
Call-ID: e12cqd3oq2kuaijqckct
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
To: sip:1001@nj.test.com;tag=c28d5129-ef12-4299-9e7c-e0c69897ed7c
CSeq: 2 INVITE
Server: FPBX-16.0.26(18.15.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Reason: Q.850;cause=34
P-Asserted-Identity: “1001” sip:1001@nj.test.com
Content-Length: 0

== Spawn extension (macro-exten-vm, s-NOANSWER, 3) exited non-zero on ‘PJSIP/1002-00000025’ in macro ‘exten-vm’
== Spawn extension (ext-local, 1001, 3) exited non-zero on ‘PJSIP/1002-00000025’
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘PJSIP/1002-00000025’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/1002-00000025’
<— Received SIP request (299 bytes) from WS:127.0.0.1:55620 —>
ACK sip:1001@nj.test.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.119;branch=z9hG4bK4556358
To: sip:1001@nj.test.com;tag=c28d5129-ef12-4299-9e7c-e0c69897ed7c
From: “Sunil K” sip:1002@nj.test.com;tag=ea4e9lnhfs
Call-ID: e12cqd3oq2kuaijqckct
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

4

503 is a catch all error. The real error is 480, and that is coming from the phone, so you need to look at the phone, rather than Asterisk. If you looked at the full definition of 480, one possibility is that it is set to do not disturb.

Also, you seem to be using FreePBX, and if this were an Asteirsk problem, you are likely to have to consider the constraints imposed by FreePBX, for which this is the wrong forum; you would want the FreePBX one.

You are also using WebRTC, and the perceived wisdom here is that you need quite a deep understanding of a substantial number of technologies in order to succeed with that. It isn’t really plug and play.

Although I’ve no reason to believe it is the problem here, you are adding further complications by running the soft phone and Asterisk on the same machine.

5

Thanks for response @david551 .

6

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