I think it is caused by network/routing. To be more specific, network configuration is shown below. A static route is set for desktop from 136.138.* to 10.94.. However, it seems traffic can’t route back from 10.94. to 136.138.* shown from the log below.
Desktop (soft phone) interface
192.168.* (internet)
136.138.* (internal network)
Asterisk interface
10.94.* (internal network)
[Aug 12 10:43:59] <— Received SIP request (1030 bytes) from UDP:136.138.1.20:57145 —>
[Aug 12 10:43:59] INVITE sip:1103@10.94.1.239 SIP/2.0
[Aug 12 10:43:59] Via: SIP/2.0/UDP 136.138.1.20:57145;rport;branch=z9hG4bKPj5a9d81a1135a48f98990b4efd0491dcc
[Aug 12 10:43:59] Max-Forwards: 70
[Aug 12 10:43:59] From: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:43:59] To: sip:1103@10.94.1.239
[Aug 12 10:43:59] Contact: “1101” sip:1101@136.138.1.20:57145;ob
[Aug 12 10:43:59] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:43:59] CSeq: 19367 INVITE
[Aug 12 10:43:59] Route: sip:10.94.1.239;lr
[Aug 12 10:43:59] Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[Aug 12 10:43:59] Supported: replaces, 100rel, timer, norefersub
[Aug 12 10:43:59] Session-Expires: 1800
[Aug 12 10:43:59] Min-SE: 90
[Aug 12 10:43:59] User-Agent: MicroSIP/3.19.8
[Aug 12 10:43:59] Content-Type: application/sdp
[Aug 12 10:43:59] Content-Length: 365
[Aug 12 10:43:59]
[Aug 12 10:43:59] v=0
[Aug 12 10:43:59] o=- 3774595843 3774595843 IN IP4 192.168.1.80
[Aug 12 10:43:59] s=pjmedia
[Aug 12 10:43:59] b=AS:84
[Aug 12 10:43:59] t=0 0
[Aug 12 10:43:59] a=X-nat:0
[Aug 12 10:43:59] m=audio 4000 RTP/AVP 0 8 3 101
[Aug 12 10:43:59] c=IN IP4 192.168.1.80
[Aug 12 10:43:59] b=TIAS:64000
[Aug 12 10:43:59] a=rtcp:4001 IN IP4 192.168.1.80
[Aug 12 10:43:59] a=sendrecv
[Aug 12 10:43:59] a=rtpmap:0 PCMU/8000
[Aug 12 10:43:59] a=rtpmap:8 PCMA/8000
[Aug 12 10:43:59] a=rtpmap:3 GSM/8000
[Aug 12 10:43:59] a=rtpmap:101 telephone-event/8000
[Aug 12 10:43:59] a=fmtp:101 0-16
[Aug 12 10:43:59] a=ssrc:802051384 cname:4bd51b97293f0aad
[Aug 12 10:43:59]
[Aug 12 10:43:59] <— Transmitting SIP response (560 bytes) to UDP:136.138.1.20:57145 —>
[Aug 12 10:43:59] SIP/2.0 401 Unauthorized
[Aug 12 10:43:59] Via: SIP/2.0/UDP 136.138.1.20:57145;rport=57145;received=136.138.1.20;branch=z9hG4bKPj5a9d81a1135a48f98990b4efd0491dcc
[Aug 12 10:43:59] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:43:59] From: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:43:59] To: sip:1103@10.94.1.239;tag=z9hG4bKPj5a9d81a1135a48f98990b4efd0491dcc
[Aug 12 10:43:59] CSeq: 19367 INVITE
[Aug 12 10:43:59] WWW-Authenticate: Digest realm=“asterisk”,nonce=“1565577839/ea6dd23a65ca96354b1bb8e0b49aefe7”,opaque=“2fb6c6c63ff2c549”,algorithm=md5,qop=“auth”
[Aug 12 10:43:59] Server: Asterisk PBX 16.5.0
[Aug 12 10:43:59] Content-Length: 0
[Aug 12 10:43:59]
[Aug 12 10:43:59]
[Aug 12 10:43:59] <— Received SIP request (409 bytes) from UDP:136.138.1.20:57145 —>
[Aug 12 10:43:59] ACK sip:1103@10.94.1.239 SIP/2.0
[Aug 12 10:43:59] Via: SIP/2.0/UDP 136.138.1.20:57145;rport;branch=z9hG4bKPj5a9d81a1135a48f98990b4efd0491dcc
[Aug 12 10:43:59] Max-Forwards: 70
[Aug 12 10:43:59] From: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:43:59] To: sip:1103@10.94.1.239;tag=z9hG4bKPj5a9d81a1135a48f98990b4efd0491dcc
[Aug 12 10:43:59] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:43:59] CSeq: 19367 ACK
[Aug 12 10:43:59] Route: sip:10.94.1.239;lr
[Aug 12 10:43:59] Content-Length: 0
[Aug 12 10:43:59]
[Aug 12 10:43:59]
[Aug 12 10:44:00] <— Received SIP request (1329 bytes) from UDP:136.138.1.20:57145 —>
[Aug 12 10:44:00] INVITE sip:1103@10.94.1.239 SIP/2.0
[Aug 12 10:44:00] Via: SIP/2.0/UDP 192.168.1.80:57145;rport;branch=z9hG4bKPj1c5121ea13ab4f329c8a325c858de2c5
[Aug 12 10:44:00] Max-Forwards: 70
[Aug 12 10:44:00] From: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:44:00] To: sip:1103@10.94.1.239
[Aug 12 10:44:00] Contact: “1101” sip:1101@136.138.1.20:57145;ob
[Aug 12 10:44:00] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:44:00] CSeq: 19368 INVITE
[Aug 12 10:44:00] Route: sip:10.94.1.239;lr
[Aug 12 10:44:00] Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[Aug 12 10:44:00] Supported: replaces, 100rel, timer, norefersub
[Aug 12 10:44:00] Session-Expires: 1800
[Aug 12 10:44:00] Min-SE: 90
[Aug 12 10:44:00] User-Agent: MicroSIP/3.19.8
[Aug 12 10:44:00] Authorization: Digest username=“3605657CFB45”, realm=“asterisk”, nonce=“1565577839/ea6dd23a65ca96354b1bb8e0b49aefe7”, uri="sip:1103@10.94.1.239", response=“95153a235e8fbc7e3b0db92a44363ea9”, algorithm=md5, cnonce=“32fecc120e104b8a8d65dbfce39c3d2b”, opaque=“2fb6c6c63ff2c549”, qop=auth, nc=00000001
[Aug 12 10:44:00] Content-Type: application/sdp
[Aug 12 10:44:00] Content-Length: 365
[Aug 12 10:44:00]
[Aug 12 10:44:00] v=0
[Aug 12 10:44:00] o=- 3774595843 3774595843 IN IP4 192.168.1.80
[Aug 12 10:44:00] s=pjmedia
[Aug 12 10:44:00] b=AS:84
[Aug 12 10:44:00] t=0 0
[Aug 12 10:44:00] a=X-nat:0
[Aug 12 10:44:00] m=audio 4000 RTP/AVP 0 8 3 101
[Aug 12 10:44:00] c=IN IP4 192.168.1.80
[Aug 12 10:44:00] b=TIAS:64000
[Aug 12 10:44:00] a=rtcp:4001 IN IP4 192.168.1.80
[Aug 12 10:44:00] a=sendrecv
[Aug 12 10:44:00] a=rtpmap:0 PCMU/8000
[Aug 12 10:44:00] a=rtpmap:8 PCMA/8000
[Aug 12 10:44:00] a=rtpmap:3 GSM/8000
[Aug 12 10:44:00] a=rtpmap:101 telephone-event/8000
[Aug 12 10:44:00] a=fmtp:101 0-16
[Aug 12 10:44:00] a=ssrc:802051384 cname:4bd51b97293f0aad
[Aug 12 10:44:00]
[Aug 12 10:44:00] == Setting global variable ‘SIPDOMAIN’ to ‘10.94.1.239’
[Aug 12 10:44:00] <— Transmitting SIP response (361 bytes) to UDP:136.138.1.20:57145 —>
[Aug 12 10:44:00] SIP/2.0 100 Trying
[Aug 12 10:44:00] Via: SIP/2.0/UDP 192.168.1.80:57145;rport=57145;received=136.138.1.20;branch=z9hG4bKPj1c5121ea13ab4f329c8a325c858de2c5
[Aug 12 10:44:00] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:44:00] From: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:44:00] To: sip:1103@10.94.1.239
[Aug 12 10:44:00] CSeq: 19368 INVITE
[Aug 12 10:44:00] Server: Asterisk PBX 16.5.0
[Aug 12 10:44:00] Content-Length: 0
[Aug 12 10:44:00]
[Aug 12 10:44:00]
[Aug 12 10:44:00] – Executing [1103@Long-Distance:1] NoOp(“PJSIP/1101-00000001”, “”) in new stack
[Aug 12 10:44:00] – Executing [1103@Long-Distance:2] Set(“PJSIP/1101-00000001”, “CDR_PROP(disable)=1”) in new stack
[Aug 12 10:44:00] – Executing [1103@Long-Distance:3] Goto(“PJSIP/1101-00000001”, “Internal-Main,1103,1”) in new stack
[Aug 12 10:44:00] – Goto (Internal-Main,1103,1)
[Aug 12 10:44:00] – Executing [1103@Internal-Main:1] Verbose(“PJSIP/1101-00000001”, “1, “User 1101 dialed 1103.””) in new stack
[Aug 12 10:44:00] “User 1101 dialed 1103.”
[Aug 12 10:44:00] – Executing [1103@Internal-Main:2] Set(“PJSIP/1101-00000001”, “SAC_DIALED_EXTEN=1103”) in new stack
[Aug 12 10:44:00] – Executing [1103@Internal-Main:3] GotoIf(“PJSIP/1101-00000001”, “0?dialed-BUSY,1:”) in new stack
[Aug 12 10:44:00] – Executing [1103@Internal-Main:4] Dial(“PJSIP/1101-00000001”, “PJSIP/1103,30”) in new stack
[Aug 12 10:44:00] ERROR[8564]: res_pjsip.c:3533 ast_sip_create_dialog_uac: Endpoint ‘1103’: Could not create dialog to invalid URI ‘1103’. Is endpoint registered and reachable?
[Aug 12 10:44:00] ERROR[8564]: chan_pjsip.c:2509 request: Failed to create outgoing session to endpoint ‘1103’
[Aug 12 10:44:00] WARNING[8569][C-00000002]: app_dial.c:2578 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
[Aug 12 10:44:00] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 12 10:44:00] – Executing [1103@Internal-Main:5] Goto(“PJSIP/1101-00000001”, “dialed-CHANUNAVAIL,1”) in new stack
[Aug 12 10:44:00] – Goto (Internal-Main,dialed-CHANUNAVAIL,1)
[Aug 12 10:44:00] – Executing [dialed-CHANUNAVAIL@Internal-Main:1] NoOp(“PJSIP/1101-00000001”, “”) in new stack
[Aug 12 10:44:00] – Executing [dialed-CHANUNAVAIL@Internal-Main:2] Playback(“PJSIP/1101-00000001”, “pbx-invalid”) in new stack
[Aug 12 10:44:00] > 0x7f57f40344d0 – Strict RTP learning after remote address set to: 192.168.1.80:4000
[Aug 12 10:44:00] <— Transmitting SIP response (915 bytes) to UDP:136.138.1.20:57145 —>
[Aug 12 10:44:00] SIP/2.0 200 OK
[Aug 12 10:44:00] Via: SIP/2.0/UDP 192.168.1.80:57145;rport=57145;received=136.138.1.20;branch=z9hG4bKPj1c5121ea13ab4f329c8a325c858de2c5
[Aug 12 10:44:00] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:44:00] From: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:44:00] To: sip:1103@10.94.1.239;tag=1fce3539-a1db-437c-8c28-41ce79130a81
[Aug 12 10:44:00] CSeq: 19368 INVITE
[Aug 12 10:44:00] Server: Asterisk PBX 16.5.0
[Aug 12 10:44:00] Contact: sip:10.94.1.239:5060
[Aug 12 10:44:00] Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
[Aug 12 10:44:00] Supported: 100rel, timer, replaces, norefersub
[Aug 12 10:44:00] Session-Expires: 1800;refresher=uac
[Aug 12 10:44:00] Require: timer
[Aug 12 10:44:00] Content-Type: application/sdp
[Aug 12 10:44:00] Content-Length: 237
[Aug 12 10:44:00]
[Aug 12 10:44:00] v=0
[Aug 12 10:44:00] o=- 3774595843 3774595845 IN IP4 10.94.1.239
[Aug 12 10:44:00] s=Asterisk
[Aug 12 10:44:00] c=IN IP4 10.94.1.239
[Aug 12 10:44:00] t=0 0
[Aug 12 10:44:00] m=audio 15544 RTP/AVP 0 101
[Aug 12 10:44:00] a=rtpmap:0 PCMU/8000
[Aug 12 10:44:00] a=rtpmap:101 telephone-event/8000
[Aug 12 10:44:00] a=fmtp:101 0-16
[Aug 12 10:44:00] a=ptime:20
[Aug 12 10:44:00] a=maxptime:150
[Aug 12 10:44:00] a=sendrecv
[Aug 12 10:44:00]
[Aug 12 10:44:00] <— Received SIP request (374 bytes) from UDP:136.138.1.20:57145 —>
[Aug 12 10:44:00] ACK sip:10.94.1.239:5060 SIP/2.0
[Aug 12 10:44:00] Via: SIP/2.0/UDP 136.138.1.20:57145;rport;branch=z9hG4bKPj7514b0543b0043aab03dcbfdbb36b77a
[Aug 12 10:44:00] Max-Forwards: 70
[Aug 12 10:44:00] From: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:44:00] To: sip:1103@10.94.1.239;tag=1fce3539-a1db-437c-8c28-41ce79130a81
[Aug 12 10:44:00] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:44:00] CSeq: 19368 ACK
[Aug 12 10:44:00] Content-Length: 0
[Aug 12 10:44:00]
[Aug 12 10:44:00]
[Aug 12 10:44:00] – <PJSIP/1101-00000001> Playing ‘pbx-invalid.ulaw’ (language ‘en’)
[Aug 12 10:44:00] > 0x7f57f40344d0 – Strict RTP qualifying stream type: audio
[Aug 12 10:44:00] > 0x7f57f40344d0 – Strict RTP switching source address to 136.138.1.20:4000
[Aug 12 10:44:04] – Executing [dialed-CHANUNAVAIL@Internal-Main:3] Hangup(“PJSIP/1101-00000001”, “”) in new stack
[Aug 12 10:44:04] == Spawn extension (Internal-Main, dialed-CHANUNAVAIL, 3) exited non-zero on ‘PJSIP/1101-00000001’
[Aug 12 10:44:04] – Executing [h@Internal-Main:1] Hangup(“PJSIP/1101-00000001”, “”) in new stack
[Aug 12 10:44:04] == Spawn extension (Internal-Main, h, 1) exited non-zero on ‘PJSIP/1101-00000001’
[Aug 12 10:44:04] <— Transmitting SIP request (442 bytes) to UDP:136.138.1.20:57145 —>
[Aug 12 10:44:04] BYE sip:1101@136.138.1.20:57145;ob SIP/2.0
[Aug 12 10:44:04] Via: SIP/2.0/UDP 10.94.1.239:5060;rport;branch=z9hG4bKPjf9cbdd01-4ed5-4901-9a2e-51249d234e09
[Aug 12 10:44:04] From: sip:1103@10.94.1.239;tag=1fce3539-a1db-437c-8c28-41ce79130a81
[Aug 12 10:44:04] To: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:44:04] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:44:04] CSeq: 25656 BYE
[Aug 12 10:44:04] Reason: Q.850;cause=3
[Aug 12 10:44:04] Max-Forwards: 70
[Aug 12 10:44:04] User-Agent: Asterisk PBX 16.5.0
[Aug 12 10:44:04] Content-Length: 0
[Aug 12 10:44:04]
[Aug 12 10:44:04]
[Aug 12 10:44:04] <— Received SIP response (365 bytes) from UDP:136.138.1.20:57145 —>
[Aug 12 10:44:04] SIP/2.0 200 OK
[Aug 12 10:44:04] Via: SIP/2.0/UDP 10.94.1.239:5060;rport=5060;received=10.94.1.239;branch=z9hG4bKPjf9cbdd01-4ed5-4901-9a2e-51249d234e09
[Aug 12 10:44:04] Call-ID: 156bb8f35113460aa473c4c16bd2627f
[Aug 12 10:44:04] From: sip:1103@10.94.1.239;tag=1fce3539-a1db-437c-8c28-41ce79130a81
[Aug 12 10:44:04] To: “1101” sip:1101@10.94.1.239;tag=99bee8163242451290937c72ad693816
[Aug 12 10:44:04] CSeq: 25656 BYE
[Aug 12 10:44:04] Content-Length: 0
[Aug 12 10:44:04]
[Aug 12 10:44:04]