Warning: 380 "SIPS not allowed"

Asterisk Asterisk SIP
1

Hi,

I am currently working on integrating Flexisip as a SIP proxy with Asterisk to handle push notifications.
The integration functions correctly over TCP and UDP protocols.
However, when attempting to use TLS, I encounter an issue where endpoints register successfully, but calls fail with the following message:

SIP/2.0 480 Temporarily Unavailable
Warning: 380 “SIPS not allowed”

I would like to inquire whether Asterisk supports the SIPS protocol. If it does, could you please provide guidance on how to configure it to work with Flexisip, which listens on SIPS:<Domain>:5061?

Thank you for your assistance.

2

It does, and you specify sips as the URI scheme wherever URIs are used instead of sip. I can’t really provide any further information as no logging, configuration, or details have been provided.

3

I have added the logs,

Asterisk 16.25.3, Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others.
Created by Mark Spencer markster@digium.com
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.

KConnected to Asterisk 16.25.3 currently running (pid = 47581)
[Mar 11 14:49:40] <— Received SIP request (1701 bytes) from TLS:Public_IP:32243 —>
[Mar 11 14:49:40] INVITE sip:4001@Domain_Name SIP/2.0
[Mar 11 14:49:40] Via: SIP/2.0/TLS Domain_Name;rport;branch=z9hG4bK.yFrppBB52S677gHD8yXagB9Qve
[Mar 11 14:49:40] Via: SIP/2.0/TLS device_IP:32930;branch=z9hG4bK.0I18VqgPN;rport=22478;received=Public_IP
[Mar 11 14:49:40] From: “4002” sip:4002@Domain_Name;tag=~obL1ICeZ
[Mar 11 14:49:40] To: sip:4001@Domain_Name
[Mar 11 14:49:40] CSeq: 20 INVITE
[Mar 11 14:49:40] Call-ID: u5DBjIe2aD
[Mar 11 14:49:40] Max-Forwards: 69
[Mar 11 14:49:40] Supported: replaces, outbound, gruu, path, record-aware
[Mar 11 14:49:40] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
[Mar 11 14:49:40] Content-Type: application/sdp
[Mar 11 14:49:40] Contact: “4002” sip:4002@Public_IP:22478;pn-prid=cruX76XhRruLxNzKT1iT3F:APA91bHd63dM0WBIT7zouPavwLJGx60Nn35WamJ3ltcqYgycVgXd_vYW0hj6sjnibvrTEPeHF2-pKgTg04yc90g0OTWRxwptR_koO5Cy_nTHqFql-IOeXcw;pn-provider=fcm;pn-param=251776523448;pn-silent=1;pn-timeout=0;transport=tls;+sip.instance=“urn:uuid:d4ebbe38-9b73-0026-8149-2b7ba98c920a”
[Mar 11 14:49:40] User-Agent: LinphoneAndroid/5.2.5 (E2info’s Tab S6 Lite) LinphoneSDK/5.3.47 (tags/5.3.47^0)
[Mar 11 14:49:40] Content-Length: 564
[Mar 11 14:49:40] Route: sip:Domain_Name:5071;transport=tls;lr
[Mar 11 14:49:40] Record-Route: sips:Domain_Name:5061;lr
[Mar 11 14:49:40]
[Mar 11 14:49:40] v=0
[Mar 11 14:49:40] o=4002 3470 2076 IN IP4 device_IP
[Mar 11 14:49:40] s=Talk
[Mar 11 14:49:40] c=IN IP4 Public_IP
[Mar 11 14:49:40] t=0 0
[Mar 11 14:49:40] a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
[Mar 11 14:49:40] a=record:off
[Mar 11 14:49:40] a=nortpproxy:yes
[Mar 11 14:49:40] m=audio 16168 RTP/AVP 96 97 98 0 8 18 101 99 100
[Mar 11 14:49:40] a=rtpmap:96 opus/48000/2
[Mar 11 14:49:40] a=fmtp:96 useinbandfec=1
[Mar 11 14:49:40] a=rtpmap:97 speex/16000
[Mar 11 14:49:40] a=fmtp:97 vbr=on
[Mar 11 14:49:40] a=rtpmap:98 speex/8000
[Mar 11 14:49:40] a=fmtp:98 vbr=on
[Mar 11 14:49:40] a=fmtp:18 annexb=yes
[Mar 11 14:49:40] a=rtpmap:101 telephone-event/48000
[Mar 11 14:49:40] a=rtpmap:99 telephone-event/16000
[Mar 11 14:49:40] a=rtpmap:100 telephone-event/8000
[Mar 11 14:49:40] a=rtcp:16169
[Mar 11 14:49:40] a=rtcp-fb:* trr-int 1000
[Mar 11 14:49:40] a=rtcp-fb:* ccm tmmbr
[Mar 11 14:49:40]

K[Mar 11 14:49:40] <— Transmitting SIP response (664 bytes) to TLS:Public_IP:32243 —>
[Mar 11 14:49:40] SIP/2.0 401 Unauthorized
[Mar 11 14:49:40] Via: SIP/2.0/TLS Domain_Name;rport=32243;received=Public_IP;branch=z9hG4bK.yFrppBB52S677gHD8yXagB9Qve
[Mar 11 14:49:40] Via: SIP/2.0/TLS device_IP:32930;rport=22478;received=Public_IP;branch=z9hG4bK.0I18VqgPN
[Mar 11 14:49:40] Record-Route: sips:Domain_Name:5061;lr
[Mar 11 14:49:40] Call-ID: u5DBjIe2aD
[Mar 11 14:49:40] From: “4002” sip:4002@Domain_Name;tag=~obL1ICeZ
[Mar 11 14:49:40] To: sip:4001@Domain_Name;tag=z9hG4bK.yFrppBB52S677gHD8yXagB9Qve
[Mar 11 14:49:40] CSeq: 20 INVITE
[Mar 11 14:49:40] WWW-Authenticate: Digest realm=“asterisk”,nonce=“1741684780/0547b51407d04f4a4d8bca2f2b6b168a”,opaque=“183dfd466ab569da”,algorithm=md5,qop=“auth”
[Mar 11 14:49:40] Server: Asterisk PBX 16.25.3
[Mar 11 14:49:40] Content-Length: 0
[Mar 11 14:49:40]
[Mar 11 14:49:40]

K[Mar 11 14:49:40] <— Received SIP request (402 bytes) from TLS:Public_IP:32243 —>
[Mar 11 14:49:40] ACK sip:4001@Domain_Name SIP/2.0
[Mar 11 14:49:40] Via: SIP/2.0/TLS Domain_Name;rport;branch=z9hG4bK.yFrppBB52S677gHD8yXagB9Qve
[Mar 11 14:49:40] Route: sip:Domain_Name:5071;transport=tls;lr
[Mar 11 14:49:40] Max-Forwards: 69
[Mar 11 14:49:40] From: “4002” sip:4002@Domain_Name;tag=~obL1ICeZ
[Mar 11 14:49:40] To: sip:4001@Domain_Name;tag=z9hG4bK.yFrppBB52S677gHD8yXagB9Qve
[Mar 11 14:49:40] Call-ID: u5DBjIe2aD
[Mar 11 14:49:40] CSeq: 20 ACK
[Mar 11 14:49:40] Content-Length: 0
[Mar 11 14:49:40]
[Mar 11 14:49:40]

K[Mar 11 14:49:40] <— Received SIP request (1986 bytes) from TLS:Public_IP:32243 —>
[Mar 11 14:49:40] INVITE sip:4001@Domain_Name SIP/2.0
[Mar 11 14:49:40] Via: SIP/2.0/TLS Domain_Name;rport;branch=z9hG4bK.5Qm1v55873Sv1XrvSj6Z1r3NKB
[Mar 11 14:49:40] Via: SIP/2.0/TLS device_IP:32930;branch=z9hG4bK.pU2-aKeRL;rport=22478;received=Public_IP
[Mar 11 14:49:40] From: “4002” sip:4002@Domain_Name;tag=~obL1ICeZ
[Mar 11 14:49:40] To: sip:4001@Domain_Name
[Mar 11 14:49:40] CSeq: 21 INVITE
[Mar 11 14:49:40] Call-ID: u5DBjIe2aD
[Mar 11 14:49:40] Max-Forwards: 69
[Mar 11 14:49:40] Supported: replaces, outbound, gruu, path, record-aware
[Mar 11 14:49:40] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
[Mar 11 14:49:40] Content-Type: application/sdp
[Mar 11 14:49:40] Contact: “4002” sip:4002@Public_IP:22478;pn-prid=cruX76XhRruLxNzKT1iT3F:APA91bHd63dM0WBIT7zouPavwLJGx60Nn35WamJ3ltcqYgycVgXd_vYW0hj6sjnibvrTEPeHF2-pKgTg04yc90g0OTWRxwptR_koO5Cy_nTHqFql-IOeXcw;pn-provider=fcm;pn-param=251776523448;pn-silent=1;pn-timeout=0;transport=tls;+sip.instance=“urn:uuid:d4ebbe38-9b73-0026-8149-2b7ba98c920a”
[Mar 11 14:49:40] User-Agent: LinphoneAndroid/5.2.5 (E2info’s Tab S6 Lite) LinphoneSDK/5.3.47 (tags/5.3.47^0)
[Mar 11 14:49:40] Authorization: Digest realm=“asterisk”, nonce=“1741684780/0547b51407d04f4a4d8bca2f2b6b168a”, algorithm=md5, opaque=“183dfd466ab569da”, username=“4002”, uri=“sip:4001@Domain_Name”, response=“a539448f7c6b0a3b021f66699b52c858”, cnonce=“aAU867ioPfuVx3mn”, nc=00000001, qop=auth
[Mar 11 14:49:40] Content-Length: 564
[Mar 11 14:49:40] Route: sip:Domain_Name:5071;transport=tls;lr
[Mar 11 14:49:40] Record-Route: sips:Domain_Name:5061;lr
[Mar 11 14:49:40]
[Mar 11 14:49:40] v=0
[Mar 11 14:49:40] o=4002 3470 2076 IN IP4 device_IP
[Mar 11 14:49:40] s=Talk
[Mar 11 14:49:40] c=IN IP4 Public_IP
[Mar 11 14:49:40] t=0 0
[Mar 11 14:49:40] a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
[Mar 11 14:49:40] a=record:off
[Mar 11 14:49:40] a=nortpproxy:yes
[Mar 11 14:49:40] m=audio 11482 RTP/AVP 96 97 98 0 8 18 101 99 100
[Mar 11 14:49:40] a=rtpmap:96 opus/48000/2
[Mar 11 14:49:40] a=fmtp:96 useinbandfec=1
[Mar 11 14:49:40] a=rtpmap:97 speex/16000
[Mar 11 14:49:40] a=fmtp:97 vbr=on
[Mar 11 14:49:40] a=rtpmap:98 speex/8000
[Mar 11 14:49:40] a=fmtp:98 vbr=on
[Mar 11 14:49:40] a=fmtp:18 annexb=yes
[Mar 11 14:49:40] a=rtpmap:101 telephone-event/48000
[Mar 11 14:49:40] a=rtpmap:99 telephone-event/16000
[Mar 11 14:49:40] a=rtpmap:100 telephone-event/8000
[Mar 11 14:49:40] a=rtcp:11483
[Mar 11 14:49:40] a=rtcp-fb:* trr-int 1000
[Mar 11 14:49:40] a=rtcp-fb:* ccm tmmbr
[Mar 11 14:49:40]

K[Mar 11 14:49:40] <— Transmitting SIP response (473 bytes) to TLS:Public_IP:32243 —>
[Mar 11 14:49:40] SIP/2.0 100 Trying
[Mar 11 14:49:40] Via: SIP/2.0/TLS Domain_Name;rport=32243;received=Public_IP;branch=z9hG4bK.5Qm1v55873Sv1XrvSj6Z1r3NKB
[Mar 11 14:49:40] Via: SIP/2.0/TLS device_IP:32930;rport=22478;received=Public_IP;branch=z9hG4bK.pU2-aKeRL
[Mar 11 14:49:40] Record-Route: sips:Domain_Name:5061;lr
[Mar 11 14:49:40] Call-ID: u5DBjIe2aD
[Mar 11 14:49:40] From: “4002” sip:4002@Domain_Name;tag=~obL1ICeZ
[Mar 11 14:49:40] To: sip:4001@Domain_Name
[Mar 11 14:49:40] CSeq: 21 INVITE
[Mar 11 14:49:40] Server: Asterisk PBX 16.25.3
[Mar 11 14:49:40] Content-Length: 0
[Mar 11 14:49:40]
[Mar 11 14:49:40]

K[Mar 11 14:49:40] – Executing [4001@pushkit:1] NoOp(“PJSIP/4002-00000002”, “PJSIP/4001 has state NOT_INUSE”) in new stack

K[Mar 11 14:49:40] – Executing [4001@pushkit:2] Set(“PJSIP/4002-00000002”, “callerNameFromHeader=“4002” sip:4002@Domain_Name;tag=~obL1ICeZ”) in new stack

K[Mar 11 14:49:40] – Executing [4001@pushkit:3] Set(“PJSIP/4002-00000002”, “callerName=4002”) in new stack

K[Mar 11 14:49:40] – Executing [4001@pushkit:4] NoOp(“PJSIP/4002-00000002”, “Caller Name of CALLERID: 4002 is: 4002”) in new stack
[Mar 11 14:49:40] – Executing [4001@pushkit:5] Set(“PJSIP/4002-00000002”, “CALLERID(name)=4002”) in new stack

K[Mar 11 14:49:40] – Executing [4001@pushkit:6] Set(“PJSIP/4002-00000002”, “caller=4002”) in new stack

K[Mar 11 14:49:40] – Executing [4001@pushkit:7] Set(“PJSIP/4002-00000002”, “callee=4001”) in new stack
[Mar 11 14:49:40] – Executing [4001@pushkit:8] Set(“PJSIP/4002-00000002”, “ringtime=30”) in new stack

K[Mar 11 14:49:40] – Executing [4001@pushkit:9] GotoIf(“PJSIP/4002-00000002”, “1?callsuccess:callbusy”) in new stack
[Mar 11 14:49:40] – Goto (pushkit,4001,10)
[Mar 11 14:49:40] – Executing [4001@pushkit:10] Goto(“PJSIP/4002-00000002”, “State,1”) in new stack

K[Mar 11 14:49:40] – Goto (pushkit,State,1)
[Mar 11 14:49:40] – Executing [State@pushkit:1] Dial(“PJSIP/4002-00000002”, “PJSIP/4001,30”) in new stack

K[Mar 11 14:49:40] – Called PJSIP/4001

K[Mar 11 14:49:40] <— Transmitting SIP request (1840 bytes) to TLS:Public_IP:5061 —>
[Mar 11 14:49:40] INVITE sips:4001@Domain_Name:5061;pn-prid=EF272DE4340:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=21b74aa62e952f3b;CtRtd187476fef2802a0=tls:Domain_Name SIP/2.0
[Mar 11 14:49:40] Via: SIP/2.0/TLS Private_IP:5071;rport;branch=z9hG4bKPj0cc49b06-cc1a-4916-b588-5e5d1d119758;alias
[Mar 11 14:49:40] From: “4002” sip:4002@Private_IP;tag=1e4fed48-0ca7-49af-b214-05713cd2346b
[Mar 11 14:49:40] To: sips:4001@Domain_Name;pn-prid=EF272DE4340:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=21b74aa62e952f3b;CtRtd187476fef2802a0=tls:Domain_Name
[Mar 11 14:49:40] Contact: sips:asterisk@Private_IP:5071;transport=TLS
[Mar 11 14:49:40] Call-ID: a74cf042-ea5b-4c2d-929e-6e36af5e5ac0
[Mar 11 14:49:40] CSeq: 7880 INVITE
[Mar 11 14:49:40] Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
[Mar 11 14:49:40] Supported: 100rel, timer, replaces, norefersub, histinfo
[Mar 11 14:49:40] Session-Expires: 1800
[Mar 11 14:49:40] Min-SE: 90
[Mar 11 14:49:40] Max-Forwards: 70
[Mar 11 14:49:40] User-Agent: Asterisk PBX 16.25.3
[Mar 11 14:49:40] Content-Type: application/sdp
[Mar 11 14:49:40] Content-Length: 263
[Mar 11 14:49:40]
[Mar 11 14:49:40] v=0
[Mar 11 14:49:40] o=- 129245756 129245756 IN IP4 Private_IP
[Mar 11 14:49:40] s=Asterisk
[Mar 11 14:49:40] c=IN IP4 Private_IP
[Mar 11 14:49:40] t=0 0
[Mar 11 14:49:40] m=audio 11942 RTP/AVP 107 8 101
[Mar 11 14:49:40] a=rtpmap:107 opus/48000/2
[Mar 11 14:49:40] a=rtpmap:8 PCMA/8000
[Mar 11 14:49:40] a=rtpmap:101 telephone-event/8000
[Mar 11 14:49:40] a=fmtp:101 0-16
[Mar 11 14:49:40] a=ptime:20
[Mar 11 14:49:40] a=maxptime:60
[Mar 11 14:49:40] a=sendrecv
[Mar 11 14:49:40]

K[Mar 11 14:49:40] <— Received SIP response (885 bytes) from TLS:Public_IP:5061 —>
[Mar 11 14:49:40] SIP/2.0 100 Trying
[Mar 11 14:49:40] Via: SIP/2.0/TLS Private_IP:5071;rport=24942;branch=z9hG4bKPj0cc49b06-cc1a-4916-b588-5e5d1d119758;alias;received=Public_IP
[Mar 11 14:49:40] Record-Route: sips:Domain_Name:5061;lr
[Mar 11 14:49:40] From: “4002” sip:4002@Private_IP;tag=1e4fed48-0ca7-49af-b214-05713cd2346b
[Mar 11 14:49:40] To: sips:4001@Domain_Name;pn-prid=EF272DE4340:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=21b74aa62e952f3b;CtRtd187476fef2802a0=tls:Domain_Name
[Mar 11 14:49:40] Call-ID: a74cf042-ea5b-4c2d-929e-6e36af5e5ac0
[Mar 11 14:49:40] CSeq: 7880 INVITE
[Mar 11 14:49:40] Server: Flexisip/2.4.0 (sofia-sip-nta/2.0)
[Mar 11 14:49:40] Content-Length: 0
[Mar 11 14:49:40]
[Mar 11 14:49:40]

K[Mar 11 14:49:40] <— Received SIP response (923 bytes) from TLS:Public_IP:5061 —>
[Mar 11 14:49:40] SIP/2.0 480 Temporarily Unavailable
[Mar 11 14:49:40] Via: SIP/2.0/TLS Private_IP:5071;rport=24942;branch=z9hG4bKPj0cc49b06-cc1a-4916-b588-5e5d1d119758;alias;received=Public_IP
[Mar 11 14:49:40] From: “4002” sip:4002@Private_IP;tag=1e4fed48-0ca7-49af-b214-05713cd2346b
[Mar 11 14:49:40] To: sips:4001@Domain_Name;pn-prid=EF272DE4340:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=21b74aa62e952f3b;CtRtd187476fef2802a0=tls:Domain_Name;tag=KHNjQ0S246QtF
[Mar 11 14:49:40] Call-ID: a74cf042-ea5b-4c2d-929e-6e36af5e5ac0
[Mar 11 14:49:40] CSeq: 7880 INVITE
[Mar 11 14:49:40] Server: Flexisip/2.4.0 (sofia-sip-nta/2.0)
[Mar 11 14:49:40] Warning: 380 Domain_Name “SIPS not allowed”
[Mar 11 14:49:40] Content-Length: 0
[Mar 11 14:49:40]
[Mar 11 14:49:40]

K[Mar 11 14:49:40] <— Transmitting SIP request (1294 bytes) to TLS:Public_IP:5061 —>
[Mar 11 14:49:40] ACK sips:4001@Domain_Name:5061;pn-prid=EF272DE4340:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=21b74aa62e952f3b;CtRtd187476fef2802a0=tls:Domain_Name SIP/2.0
[Mar 11 14:49:40] Via: SIP/2.0/TLS Private_IP:5071;rport;branch=z9hG4bKPj0cc49b06-cc1a-4916-b588-5e5d1d119758;alias
[Mar 11 14:49:40] From: “4002” sip:4002@Private_IP;tag=1e4fed48-0ca7-49af-b214-05713cd2346b
[Mar 11 14:49:40] To: sips:4001@Domain_Name;pn-prid=EF272DE4340:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=21b74aa62e952f3b;CtRtd187476fef2802a0=tls:Domain_Name;tag=KHNjQ0S246QtF
[Mar 11 14:49:40] Call-ID: a74cf042-ea5b-4c2d-929e-6e36af5e5ac0
[Mar 11 14:49:40] CSeq: 7880 ACK
[Mar 11 14:49:40] Max-Forwards: 70
[Mar 11 14:49:40] User-Agent: Asterisk PBX 16.25.3
[Mar 11 14:49:40] Content-Length: 0
[Mar 11 14:49:40]
[Mar 11 14:49:40]

K[Mar 11 14:49:40] – No one is available to answer at this time (1:0/0/0)

K[Mar 11 14:49:40] – Executing [State@pushkit:2] Goto(“PJSIP/4002-00000002”, “h,1”) in new stack

K[Mar 11 14:49:40] – Goto (pushkit,h,1)
[Mar 11 14:49:40] – Executing [h@pushkit:1] NoOp(“PJSIP/4002-00000002”, “Dial Status is NOANSWER”) in new stack
[Mar 11 14:49:40] – Executing [h@pushkit:2] NoOp(“PJSIP/4002-00000002”, “Hangup cause is 19”) in new stack

K[Mar 11 14:49:40] – Executing [h@pushkit:3] Hangup(“PJSIP/4002-00000002”, “”) in new stack
[Mar 11 14:49:40] == Spawn extension (pushkit, h, 3) exited non-zero on ‘PJSIP/4002-00000002’

K[Mar 11 14:49:40] – Executing [h@pushkit:1] NoOp(“PJSIP/4002-00000002”, “Dial Status is NOANSWER”) in new stack

K[Mar 11 14:49:40] – Executing [h@pushkit:2] NoOp(“PJSIP/4002-00000002”, “Hangup cause is 19”) in new stack

K[Mar 11 14:49:40] – Executing [h@pushkit:3] Hangup(“PJSIP/4002-00000002”, “”) in new stack

K[Mar 11 14:49:40] == Spawn extension (pushkit, h, 3) exited non-zero on ‘PJSIP/4002-00000002’

K[Mar 11 14:49:40] <— Transmitting SIP response (555 bytes) to TLS:Public_IP:32243 —>
[Mar 11 14:49:40] SIP/2.0 480 Temporarily Unavailable
[Mar 11 14:49:40] Via: SIP/2.0/TLS Domain_Name;rport=32243;received=Public_IP;branch=z9hG4bK.5Qm1v55873Sv1XrvSj6Z1r3NKB
[Mar 11 14:49:40] Via: SIP/2.0/TLS device_IP:32930;rport=22478;received=Public_IP;branch=z9hG4bK.pU2-aKeRL
[Mar 11 14:49:40] Record-Route: sips:Domain_Name:5061;lr
[Mar 11 14:49:40] Call-ID: u5DBjIe2aD
[Mar 11 14:49:40] From: “4002” sip:4002@Domain_Name;tag=~obL1ICeZ
[Mar 11 14:49:40] To: sip:4001@Domain_Name;tag=5815c53a-3813-43e3-96a4-369f3a03c898
[Mar 11 14:49:40] CSeq: 21 INVITE
[Mar 11 14:49:40] Server: Asterisk PBX 16.25.3
[Mar 11 14:49:40] Reason: Q.850;cause=19
[Mar 11 14:49:40] Content-Length: 0
[Mar 11 14:49:40]
[Mar 11 14:49:40]

K[Mar 11 14:49:40] <— Received SIP request (404 bytes) from TLS:Public_IP:32243 —>
[Mar 11 14:49:40] ACK sip:4001@Domain_Name SIP/2.0
[Mar 11 14:49:40] Via: SIP/2.0/TLS Domain_Name;rport;branch=z9hG4bK.5Qm1v55873Sv1XrvSj6Z1r3NKB
[Mar 11 14:49:40] Route: sip:Domain_Name:5071;transport=tls;lr
[Mar 11 14:49:40] Max-Forwards: 69
[Mar 11 14:49:40] From: “4002” sip:4002@Domain_Name;tag=~obL1ICeZ
[Mar 11 14:49:40] To: sip:4001@Domain_Name;tag=5815c53a-3813-43e3-96a4-369f3a03c898
[Mar 11 14:49:40] Call-ID: u5DBjIe2aD
[Mar 11 14:49:40] CSeq: 21 ACK
[Mar 11 14:49:40] Content-Length: 0
[Mar 11 14:49:40]
[Mar 11 14:49:40]

4

What is the Contact for endpoint 4001?

Additionally, you are using an unsupported version of Asterisk. It no longer receives security fixes fyi.

5

Oh, and what is the REGISTER exchange for it?

6

It is flexsip, not Asterisk, which is rejecting the SIPS URI.

7

Here is the REGISTER capture

Asterisk 16.25.3, Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others.

Created by Mark Spencer markster@digium.com

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 16.25.3 currently running on (pid = 2963)

catie*CLI>

catie*CLI> pjsip set logger on

PJSIP Logging enabled

catie*CLI>

[Mar 11 16:51:22] <— Received SIP request (1522 bytes) from TLS:Public_IP:48000 —>

[Mar 11 16:51:22] REGISTER sip:Domain_Name SIP/2.0

[Mar 11 16:51:22] Via: SIP/2.0/TLS Domain_Name;rport;branch=z9hG4bK.7Kc16BDHK6Kvv3etjr2X08DF1g

[Mar 11 16:51:22] Path: sips:Domain_Name:5061;fs-proxy-id=d187476fef2802a0;lr

[Mar 11 16:51:22] Path: sips:Domain_Name:5061;lr

[Mar 11 16:51:22] Via: SIP/2.0/TLS device_IP:33230;alias;branch=z9hG4bK.kiZX0QJY~;rport=53221;received=Public_IP

[Mar 11 16:51:22] Max-Forwards: 69

[Mar 11 16:51:22] From: “4002” sip:4002@Domain_Name;tag=S8Rfpuiqu

[Mar 11 16:51:22] To: “4002” sip:4002@Domain_Name

[Mar 11 16:51:22] Call-ID: rREMJ3c35N

[Mar 11 16:51:22] CSeq: 66 REGISTER

[Mar 11 16:51:22] Contact: “4002” sips:4002@Domain_Name:5061;pn-prid=cruX76XhRruLxNzKT1iT3F:APA91bHd63dM0WBIT7zouPavwLJGx60Nn35WamJ3ltcqYgycVgXd_vYW0hj6sjnibvrTEPeHF2-pKgTg04yc90g0OTWRxwptR_koO5Cy_nTHqFql-IOeXcw;pn-provider=fcm;pn-param=251776523448;pn-silent=1;pn-timeout=0;fs-conn-id=1b7fd7c014b11a8f;CtRtd187476fef2802a0=tls:Domain_Name;+sip.instance=“urn:uuid:d4ebbe38-9b73-0026-8149-2b7ba98c920a”

[Mar 11 16:51:22] Expires: 3600

[Mar 11 16:51:22] User-Agent: LinphoneAndroid/5.2.5 (E2info’s Tab S6 Lite) LinphoneSDK/5.3.47 (tags/5.3.47^0)

[Mar 11 16:51:22] Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml

[Mar 11 16:51:22] Supported: replaces, outbound, gruu, path, record-aware

[Mar 11 16:51:22] Authorization: Digest realm=“asterisk”, nonce=“1741692040/d63db1103721c5f4f3fa14a9ec2d33cd”, algorithm=md5, opaque=“41f98ded1b79ac85”, username=“4002”, uri=“sip:Domain_Name”, response=“4fe43e8b56845bec41e327e87bba8464”, cnonce=“aLByK0tDiE5vmwLj”, nc=00000002, qop=auth

[Mar 11 16:51:22] Content-Length: 0

[Mar 11 16:51:22] Route: sip:Domain_Name:5071;transport=tls;lr

[Mar 11 16:51:22]

[Mar 11 16:51:22]

[Mar 11 16:51:22] <— Transmitting SIP response (638 bytes) to TLS:Public_IP:48000 —>

[Mar 11 16:51:22] SIP/2.0 401 Unauthorized

[Mar 11 16:51:22] Via: SIP/2.0/TLS Domain_Name;rport=48000;received=Public_IP;branch=z9hG4bK.7Kc16BDHK6Kvv3etjr2X08DF1g

[Mar 11 16:51:22] Via: SIP/2.0/TLS device_IP:33230;rport=53221;received=Public_IP;branch=z9hG4bK.kiZX0QJY~;alias

[Mar 11 16:51:22] Call-ID: rREMJ3c35N

[Mar 11 16:51:22] From: “4002” sip:4002@Domain_Name;tag=S8Rfpuiqu

[Mar 11 16:51:22] To: “4002” sip:4002@Domain_Name;tag=z9hG4bK.7Kc16BDHK6Kvv3etjr2X08DF1g

[Mar 11 16:51:22] CSeq: 66 REGISTER

[Mar 11 16:51:22] WWW-Authenticate: Digest realm=“asterisk”,nonce=“1741692082/e0d84e6f9a552f0a140c78b00cb04d01”,opaque=“3ecf20d24867d15e”,stale=true,algorithm=md5,qop=“auth”

[Mar 11 16:51:22] Server: Asterisk PBX 16.25.3

[Mar 11 16:51:22] Content-Length: 0

[Mar 11 16:51:22]

[Mar 11 16:51:22]

[Mar 11 16:51:22] <— Received SIP request (1522 bytes) from TLS:Public_IP:48000 —>

[Mar 11 16:51:22] REGISTER sip:Domain_Name SIP/2.0

[Mar 11 16:51:22] Via: SIP/2.0/TLS Domain_Name;rport;branch=z9hG4bK.eey440vQQXcmy1D9Z50004cFre

[Mar 11 16:51:22] Path: sips:Domain_Name:5061;fs-proxy-id=d187476fef2802a0;lr

[Mar 11 16:51:22] Path: sips:Domain_Name:5061;lr

[Mar 11 16:51:22] Via: SIP/2.0/TLS device_IP:33230;alias;branch=z9hG4bK.4Za8khZw-;rport=53221;received=Public_IP

[Mar 11 16:51:22] Max-Forwards: 69

[Mar 11 16:51:22] From: “4002” sip:4002@Domain_Name;tag=S8Rfpuiqu

[Mar 11 16:51:22] To: “4002” sip:4002@Domain_Name

[Mar 11 16:51:22] Call-ID: rREMJ3c35N

[Mar 11 16:51:22] CSeq: 67 REGISTER

[Mar 11 16:51:22] Contact: “4002” sips:4002@Domain_Name:5061;pn-prid=cruX76XhRruLxNzKT1iT3F:APA91bHd63dM0WBIT7zouPavwLJGx60Nn35WamJ3ltcqYgycVgXd_vYW0hj6sjnibvrTEPeHF2-pKgTg04yc90g0OTWRxwptR_koO5Cy_nTHqFql-IOeXcw;pn-provider=fcm;pn-param=251776523448;pn-silent=1;pn-timeout=0;fs-conn-id=1b7fd7c014b11a8f;CtRtd187476fef2802a0=tls:Domain_Name;+sip.instance=“urn:uuid:d4ebbe38-9b73-0026-8149-2b7ba98c920a”

[Mar 11 16:51:22] Expires: 3600

[Mar 11 16:51:22] User-Agent: LinphoneAndroid/5.2.5 (E2info’s Tab S6 Lite) LinphoneSDK/5.3.47 (tags/5.3.47^0)

[Mar 11 16:51:22] Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml

[Mar 11 16:51:22] Supported: replaces, outbound, gruu, path, record-aware

[Mar 11 16:51:22] Authorization: Digest realm=“asterisk”, nonce=“1741692082/e0d84e6f9a552f0a140c78b00cb04d01”, algorithm=md5, opaque=“3ecf20d24867d15e”, username=“4002”, uri=“sip:Domain_Name”, response=“4d64f6c488d97a9bf840c0e774b171b2”, cnonce=“gBxcgGcOkxPbmoIi”, nc=00000001, qop=auth

[Mar 11 16:51:22] Content-Length: 0

[Mar 11 16:51:22] Route: sip:Domain_Name:5071;transport=tls;lr

[Mar 11 16:51:22]

[Mar 11 16:51:22]

[Mar 11 16:51:22] <— Transmitting SIP response (874 bytes) to TLS:Public_IP:48000 —>

[Mar 11 16:51:22] SIP/2.0 200 OK

[Mar 11 16:51:22] Via: SIP/2.0/TLS Domain_Name;rport=48000;received=Public_IP;branch=z9hG4bK.eey440vQQXcmy1D9Z50004cFre

[Mar 11 16:51:22] Via: SIP/2.0/TLS device_IP:33230;rport=53221;received=Public_IP;branch=z9hG4bK.4Za8khZw-;alias

[Mar 11 16:51:22] Call-ID: rREMJ3c35N

[Mar 11 16:51:22] From: “4002” sip:4002@Domain_Name;tag=S8Rfpuiqu

[Mar 11 16:51:22] To: “4002” sip:4002@Domain_Name;tag=z9hG4bK.eey440vQQXcmy1D9Z50004cFre

[Mar 11 16:51:22] CSeq: 67 REGISTER

[Mar 11 16:51:22] Date: Tue, 11 Mar 2025 11:21:22 GMT

[Mar 11 16:51:22] Contact: sips:4002@Domain_Name:5061;pn-prid=cruX76XhRruLxNzKT1iT3F:APA91bHd63dM0WBIT7zouPavwLJGx60Nn35WamJ3ltcqYgycVgXd_vYW0hj6sjnibvrTEPeHF2-pKgTg04yc90g0OTWRxwptR_koO5Cy_nTHqFql-IOeXcw;pn-provider=fcm;pn-param=251776523448;pn-silent=1;pn-timeout=0;fs-conn-id=1b7fd7c014b11a8f;CtRtd187476fef2802a0=tls:Domain_Name;expires=3599

[Mar 11 16:51:22] Expires: 3600

[Mar 11 16:51:22] Server: Asterisk PBX 16.25.3

[Mar 11 16:51:22] Content-Length: 0

[Mar 11 16:51:22]

[Mar 11 16:51:22]

[Mar 11 16:51:27] <— Received SIP request (1660 bytes) from TLS:Public_IP:48000 —>

[Mar 11 16:51:27] REGISTER sip:Domain_Name SIP/2.0

[Mar 11 16:51:27] Via: SIP/2.0/TLS Domain_Name;rport;branch=z9hG4bK.2r67em8tUN8KHgFy5yNF18a9gH

[Mar 11 16:51:27] Path: sips:Domain_Name:5061;fs-proxy-id=d187476fef2802a0;lr

[Mar 11 16:51:27] Path: sips:Domain_Name:5061;lr

[Mar 11 16:51:27] Via: SIP/2.0/TLS device_IP:49349;alias;branch=z9hG4bK.~KoI3mjce;rport=35708;received=Public_IP

[Mar 11 16:51:27] Max-Forwards: 69

[Mar 11 16:51:27] From: “4001” sip:4001@Domain_Name;tag=t-ZS~cDhx

[Mar 11 16:51:27] To: “4001” sip:4001@Domain_Name

[Mar 11 16:51:27] Call-ID: BqXeSedIyt

[Mar 11 16:51:27] CSeq: 26 REGISTER

[Mar 11 16:51:27] Contact: “4001” sips:4001@Domain_Name:5061;pn-prid=EF272DE43400E79475C2494E5C1860330D8859CCA32956C247929DD389CD28F2:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=db0984b14acb6c6d;CtRtd187476fef2802a0=tls:Domain_Name;+sip.instance=“urn:uuid:18a32d35-244a-4e9f-b73a-cb0520174f9d”;+org.linphone.specs=“lime”

[Mar 11 16:51:27] Expires: 3600

[Mar 11 16:51:27] User-Agent: LinphoneiOS/5.2.4 (iPad) LinphoneSDK/5.3.89

[Mar 11 16:51:27] Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml

[Mar 11 16:51:27] Supported: replaces, outbound, gruu, path, record-aware

[Mar 11 16:51:27] Authorization: Digest realm=“asterisk”, nonce=“1741691954/66a7339a35031da893c63f099e88e02d”, algorithm=md5, opaque=“37578e6c611b13ed”, username=“4001”, uri=“sip:Domain_Name”, response=“d985db02394a1e69b122680374bbce3e”, cnonce=“ILW9D5K7UVShDEjd”, nc=00000004, qop=auth

[Mar 11 16:51:27] Content-Length: 0

[Mar 11 16:51:27] Route: sip:Domain_Name:5071;transport=tls;lr

[Mar 11 16:51:27]

[Mar 11 16:51:27]

[Mar 11 16:51:27] <— Transmitting SIP response (638 bytes) to TLS:Public_IP:48000 —>

[Mar 11 16:51:27] SIP/2.0 401 Unauthorized

[Mar 11 16:51:27] Via: SIP/2.0/TLS Domain_Name;rport=48000;received=Public_IP;branch=z9hG4bK.2r67em8tUN8KHgFy5yNF18a9gH

[Mar 11 16:51:27] Via: SIP/2.0/TLS device_IP:49349;rport=35708;received=Public_IP;branch=z9hG4bK.~KoI3mjce;alias

[Mar 11 16:51:27] Call-ID: BqXeSedIyt

[Mar 11 16:51:27] From: “4001” sip:4001@Domain_Name;tag=t-ZS~cDhx

[Mar 11 16:51:27] To: “4001” sip:4001@Domain_Name;tag=z9hG4bK.2r67em8tUN8KHgFy5yNF18a9gH

[Mar 11 16:51:27] CSeq: 26 REGISTER

[Mar 11 16:51:27] WWW-Authenticate: Digest realm=“asterisk”,nonce=“1741692087/dcfc9ba45b7881670fd98f9babd7faa8”,opaque=“3ae64b921d30eb18”,stale=true,algorithm=md5,qop=“auth”

[Mar 11 16:51:27] Server: Asterisk PBX 16.25.3

[Mar 11 16:51:27] Content-Length: 0

[Mar 11 16:51:27]

[Mar 11 16:51:27]

[Mar 11 16:51:27] <— Received SIP request (1660 bytes) from TLS:Public_IP:48000 —>

[Mar 11 16:51:27] REGISTER sip:Domain_Name SIP/2.0

[Mar 11 16:51:27] Via: SIP/2.0/TLS Domain_Name;rport;branch=z9hG4bK.g22S0pc6vH5j3cp356tmgejmDa

[Mar 11 16:51:27] Path: sips:Domain_Name:5061;fs-proxy-id=d187476fef2802a0;lr

[Mar 11 16:51:27] Path: sips:Domain_Name:5061;lr

[Mar 11 16:51:27] Via: SIP/2.0/TLS device_IP:49349;alias;branch=z9hG4bK.j~PvGH-lr;rport=35708;received=Public_IP

[Mar 11 16:51:27] Max-Forwards: 69

[Mar 11 16:51:27] From: “4001” sip:4001@Domain_Name;tag=t-ZS~cDhx

[Mar 11 16:51:27] To: “4001” sip:4001@Domain_Name

[Mar 11 16:51:27] Call-ID: BqXeSedIyt

[Mar 11 16:51:27] CSeq: 27 REGISTER

[Mar 11 16:51:27] Contact: “4001” sips:4001@Domain_Name:5061;pn-prid=EF272DE43400E79475C2494E5C1860330D8859CCA32956C247929DD389CD28F2:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=db0984b14acb6c6d;CtRtd187476fef2802a0=tls:Domain_Name;+sip.instance=“urn:uuid:18a32d35-244a-4e9f-b73a-cb0520174f9d”;+org.linphone.specs=“lime”

[Mar 11 16:51:27] Expires: 3600

[Mar 11 16:51:27] User-Agent: LinphoneiOS/5.2.4 (iPad) LinphoneSDK/5.3.89

[Mar 11 16:51:27] Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml

[Mar 11 16:51:27] Supported: replaces, outbound, gruu, path, record-aware

[Mar 11 16:51:27] Authorization: Digest realm=“asterisk”, nonce=“1741692087/dcfc9ba45b7881670fd98f9babd7faa8”, algorithm=md5, opaque=“3ae64b921d30eb18”, username=“4001”, uri=“sip:Domain_Name”, response=“4d6a17825c8ecba2c3bb1e053d0303df”, cnonce=“nXHi9MIzYbKfjOqz”, nc=00000001, qop=auth

[Mar 11 16:51:27] Content-Length: 0

[Mar 11 16:51:27] Route: sip:Domain_Name:5071;transport=tls;lr

[Mar 11 16:51:27]

[Mar 11 16:51:27]

[Mar 11 16:51:27] <— Transmitting SIP response (1021 bytes) to TLS:Public_IP:48000 —>

[Mar 11 16:51:27] SIP/2.0 200 OK

[Mar 11 16:51:27] Via: SIP/2.0/TLS Domain_Name;rport=48000;received=Public_IP;branch=z9hG4bK.g22S0pc6vH5j3cp356tmgejmDa

[Mar 11 16:51:27] Via: SIP/2.0/TLS device_IP:49349;rport=35708;received=Public_IP;branch=z9hG4bK.j~PvGH-lr;alias

[Mar 11 16:51:27] Call-ID: BqXeSedIyt

[Mar 11 16:51:27] From: “4001” sip:4001@Domain_Name;tag=t-ZS~cDhx

[Mar 11 16:51:27] To: “4001” sip:4001@Domain_Name;tag=z9hG4bK.g22S0pc6vH5j3cp356tmgejmDa

[Mar 11 16:51:27] CSeq: 27 REGISTER

[Mar 11 16:51:27] Date: Tue, 11 Mar 2025 11:21:27 GMT

[Mar 11 16:51:27] Contact: sips:4001@Domain_Name:5061;pn-prid=EF272DE43400E79475C2494E5C1860330D8859CCA32956C247929DD389CD28F2:voip&AD208ED5E666BB02754232DA283BA4FB48CE0FA86A186F013EDC653DCEE6C8AB:remote;pn-provider=apns;pn-param=ABCD1234.org.linphone.phone.voip&remote;pn-silent=1;pn-timeout=0;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-groupchat-str=GC_MSG;pn-call-snd=notes_of_the_optimistic.caf;pn-msg-snd=msg.caf;fs-conn-id=db0984b14acb6c6d;CtRtd187476fef2802a0=tls:Domain_Name;expires=3599

[Mar 11 16:51:27] Expires: 3600

[Mar 11 16:51:27] Server: Asterisk PBX 16.25.3

[Mar 11 16:51:27] Content-Length: 0

[Mar 11 16:51:27]

[Mar 11 16:51:27]

catie*CLI>

8

The REGISTER has provided a Contact with “sips” as the URI scheme. We therefore use it when dialling the endpoint, including in some of our own URIs. If that were to be “sip” instead then all the “sips” should go away. That’s not something configurable in Asterisk.

9

Thanks