This is what happens when I press the “transfer call”/Hold button on the phone from the SIP messages perspective, until the call is terminated:
[2024-02-05 00:24:50.353] <--- Received SIP request (1200 bytes) from TLS:10.25.12.14:52057 --->
[2024-02-05 00:24:50.353] INVITE sips:0100000000@10.24.12.58:5061;transport=TLS SIP/2.0
[2024-02-05 00:24:50.353] Via: SIP/2.0/TLS 10.25.12.14:52057;rport;branch=z9hG4bKPjf247e4b4-c462-47c1-929f-9448423951c5;alias
[2024-02-05 00:24:50.353] Max-Forwards: 70
[2024-02-05 00:24:50.353] From: <sips:234@10.25.12.14>;tag=1eeefad2-d4c3-4453-9687-823549430280
[2024-02-05 00:24:50.353] To: <sip:012345678901@10.24.12.58>;tag=c39995b5-df11-42bd-9f04-e84e0dbc0fd4
[2024-02-05 00:24:50.353] Contact: <sips:234@10.25.12.14:5061;transport=tls>
[2024-02-05 00:24:50.353] Call-ID: 331bfab4-44b5-4625-a33b-33b5d72cb5da
[2024-02-05 00:24:50.353] CSeq: 32611 INVITE
[2024-02-05 00:24:50.353] Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, PRACK, REFER, MESSAGE
[2024-02-05 00:24:50.353] Supported: 100rel, replaces, timer, eventlist
[2024-02-05 00:24:50.353] Session-Expires: 1800;refresher=uas
[2024-02-05 00:24:50.353] Min-SE: 90
[2024-02-05 00:24:50.353] User-Agent: elmeg IP630/82.3.19.1-release;589EC66BDF2D
[2024-02-05 00:24:50.353] Allow-Events: hold,talk
[2024-02-05 00:24:50.353] Content-Type: application/sdp
[2024-02-05 00:24:50.353] Content-Length: 419
[2024-02-05 00:24:50.353]
[2024-02-05 00:24:50.353] v=0
[2024-02-05 00:24:50.353] o=- 3916077851 3916077853 IN IP4 10.25.12.14
[2024-02-05 00:24:50.353] s=elmeg IP630/82.3.19.1-release;589EC66BDF2D
[2024-02-05 00:24:50.353] t=0 0
[2024-02-05 00:24:50.353] m=audio 8046 RTP/SAVP 9 8 0 101
[2024-02-05 00:24:50.354] c=IN IP4 10.25.12.14
[2024-02-05 00:24:50.354] b=TIAS:64000
[2024-02-05 00:24:50.354] a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:3jAXcQNdabSJSWB/kRGbATfPEa/jHIAD9fPJQihH
[2024-02-05 00:24:50.354] a=rtcp:8047 IN IP4 10.25.12.14
[2024-02-05 00:24:50.354] a=rtpmap:9 G722/8000
[2024-02-05 00:24:50.354] a=rtpmap:8 PCMA/8000
[2024-02-05 00:24:50.354] a=rtpmap:0 PCMU/8000
[2024-02-05 00:24:50.354] a=rtpmap:101 telephone-event/8000
[2024-02-05 00:24:50.354] a=fmtp:101 0-16
[2024-02-05 00:24:50.354] a=sendonly
[2024-02-05 00:24:50.354]
[2024-02-05 00:24:50.355] <--- Transmitting SIP response (1064 bytes) to TLS:10.25.12.14:52057 --->
[2024-02-05 00:24:50.355] SIP/2.0 200 OK
[2024-02-05 00:24:50.355] Via: SIP/2.0/TLS 10.25.12.14:52057;rport=52057;received=10.25.12.14;branch=z9hG4bKPjf247e4b4-c462-47c1-929f-9448423951c5;alias
[2024-02-05 00:24:50.355] Call-ID: 331bfab4-44b5-4625-a33b-33b5d72cb5da
[2024-02-05 00:24:50.355] From: <sips:234@10.25.12.14>;tag=1eeefad2-d4c3-4453-9687-823549430280
[2024-02-05 00:24:50.355] To: <sip:012345678901@10.24.12.58>;tag=c39995b5-df11-42bd-9f04-e84e0dbc0fd4
[2024-02-05 00:24:50.355] CSeq: 32611 INVITE
[2024-02-05 00:24:50.355] Session-Expires: 1800;refresher=uas
[2024-02-05 00:24:50.356] Contact: <sip:0100000000@10.24.12.58:5061;transport=TLS>
[2024-02-05 00:24:50.356] Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
[2024-02-05 00:24:50.356] Supported: 100rel, timer, replaces, norefersub
[2024-02-05 00:24:50.356] Server: Asterisk PBX
[2024-02-05 00:24:50.356] Content-Type: application/sdp
[2024-02-05 00:24:50.356] Content-Length: 366
[2024-02-05 00:24:50.356]
[2024-02-05 00:24:50.356] v=0
[2024-02-05 00:24:50.356] o=- 511053411 511053412 IN IP4 10.24.12.58
[2024-02-05 00:24:50.356] s=Asterisk
[2024-02-05 00:24:50.356] c=IN IP4 10.24.12.58
[2024-02-05 00:24:50.357] t=0 0
[2024-02-05 00:24:50.357] m=audio 15248 RTP/SAVP 9 8 0 101
[2024-02-05 00:24:50.357] a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:YTvDRwsfUxy9tljBWO2EHacLr+23NowN1G+gzMK2
[2024-02-05 00:24:50.357] a=rtpmap:9 G722/8000
[2024-02-05 00:24:50.357] a=rtpmap:8 PCMA/8000
[2024-02-05 00:24:50.357] a=rtpmap:0 PCMU/8000
[2024-02-05 00:24:50.357] a=rtpmap:101 telephone-event/8000
[2024-02-05 00:24:50.357] a=fmtp:101 0-16
[2024-02-05 00:24:50.357] a=ptime:20
[2024-02-05 00:24:50.357] a=maxptime:140
[2024-02-05 00:24:50.357] a=recvonly
[2024-02-05 00:24:50.358]
[2024-02-05 00:24:50.358] -- Started music on hold, class 'default', on channel 'PJSIP/mytrunk-00000002'
[2024-02-05 00:24:50.393] <--- Received SIP request (449 bytes) from TLS:10.25.12.14:52057 --->
[2024-02-05 00:24:50.393] BYE sip:0100000000@10.24.12.58:5061;transport=TLS SIP/2.0
[2024-02-05 00:24:50.394] Via: SIP/2.0/TLS 10.25.12.14:52057;rport;branch=z9hG4bKPj98d61745-61c7-4bfe-a62a-375b7b9ed121;alias
[2024-02-05 00:24:50.394] Max-Forwards: 70
[2024-02-05 00:24:50.394] From: <sips:234@10.25.12.14>;tag=1eeefad2-d4c3-4453-9687-823549430280
[2024-02-05 00:24:50.394] To: <sip:012345678901@10.24.12.58>;tag=c39995b5-df11-42bd-9f04-e84e0dbc0fd4
[2024-02-05 00:24:50.394] Call-ID: 331bfab4-44b5-4625-a33b-33b5d72cb5da
[2024-02-05 00:24:50.395] CSeq: 32612 BYE
[2024-02-05 00:24:50.395] Warning: 381 max3b "SIPS Required"
[2024-02-05 00:24:50.395] Content-Length: 0
[2024-02-05 00:24:50.395]
[2024-02-05 00:24:50.395]
[2024-02-05 00:24:50.395] <--- Transmitting SIP response (400 bytes) to TLS:10.25.12.14:52057 --->
[2024-02-05 00:24:50.396] SIP/2.0 200 OK
[2024-02-05 00:24:50.396] Via: SIP/2.0/TLS 10.25.12.14:52057;rport=52057;received=10.25.12.14;branch=z9hG4bKPj98d61745-61c7-4bfe-a62a-375b7b9ed121;alias
[2024-02-05 00:24:50.396] Call-ID: 331bfab4-44b5-4625-a33b-33b5d72cb5da
[2024-02-05 00:24:50.396] From: <sips:234@10.25.12.14>;tag=1eeefad2-d4c3-4453-9687-823549430280
[2024-02-05 00:24:50.396] To: <sip:012345678901@10.24.12.58>;tag=c39995b5-df11-42bd-9f04-e84e0dbc0fd4
[2024-02-05 00:24:50.396] CSeq: 32612 BYE
[2024-02-05 00:24:50.396] Server: Asterisk PBX
[2024-02-05 00:24:50.396] Content-Length: 0
[2024-02-05 00:24:50.397]
[2024-02-05 00:24:50.397]
[2024-02-05 00:24:50.397] -- Channel PJSIP/234-00000003 left 'simple_bridge' basic-bridge <17df271c-5bcc-4b89-8a65-802e7bad0661>
[2024-02-05 00:24:50.397] -- Channel PJSIP/mytrunk-00000002 left 'simple_bridge' basic-bridge <17df271c-5bcc-4b89-8a65-802e7bad0661>
[2024-02-05 00:24:50.398] -- Stopped music on hold on PJSIP/mytrunk-00000002
[2024-02-05 00:24:50.398] == Spawn extension (mytrunk, 4910000000234, 1) exited non-zero on 'PJSIP/mytrunk-00000002'
Any ideas about
Warning: 381 max3b "SIPS Required"
in the phone’s message?
It seems the phone (10.25.12.14
) says BYE
, maybe because it didn’t like Asterisk’s (10.24.12.58
) response? Everything wents on so quick that I cannot hear any music on hold before the hangup.
In the traces, 012345678901
is the national number of an external caller whose call I tried to redirect to a colleague’s phone.