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Ruisheng Peng

2021-Feb-06 01:29 UTC

[asterisk-users] Hangup() not working for handsets using pls transport?

Thanks Jashua for the suggestion.  To find out if the issue was only
limited to the softphone that was using tls transport (SOFTPHONE_B on ext
103, a linphone running off my MBP), I also turned one of the hard phone
(0000f30A0A01 on ext 100, a Yealink T32G) into using tls transport.  It
behaves similarly to the linphone in that the Hangup() call in dialplan is
silently ignored, and the handsets would alway appear as busy/unavilable.

Here're the relevant part of my /etc/asterisk/extensions.conf:

[globals]

; General internal dialing options used in context Dial-Users.

; Only the timeout is defined here. See the Dial app documentation for

; additional options.

INTERNAL_DIAL_OPT=,30

RP_Yealink = PJSIP/0000f30A0A01

RP_Cisco = PJSIP/0000f30B0B02

RP_HMBP = PJSIP/SOFTPHONE_A

RP_OMBP = PJSIP/SOFTPHONE_B


[sets]

exten => 100,1,Dial(${RP_Yealink},10,m)

        same => n,Playback(vm-nobodyavail)

        same => n,Hangup()


exten => 101,1,Dial(${RP_Cisco},10)

        same => n,Playback(vm-nobodyavail)

        same => n,Hangup()


exten => 102,1,Dial(${RP_HMBP})


exten => 103,1,Dial(${RP_OMBP},10)

        same => n,Playback(vm-nobodyavail)

        same => n,Hangup()


exten => 110,1,Dial(${RP_Yealink}&${RP_Cisco})


exten => 200,1,Answer()

   same => n,Playback(hello-world)

   same => n,Hangup()

  Here're what pjsip logger captures when using the tls softphone (on ext
103) to call ext 101 (Hello World!). I had to click the hanup button on the
linphone some 15s later to terminate the call.

<--- Received SIP request (1199 bytes) from UDP:128.171.168.233:5060 --->

INVITE sip:200 at 128.171.77.23 SIP/2.0

Via: SIP/2.0/UDP 128.171.168.233:5060;branch=z9hG4bK.D-YbrxKYs;rport

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: sip:200 at 128.171.77.23

CSeq: 20 INVITE

Call-ID: ziUzVUxYw7

Max-Forwards: 70

Supported: replaces, outbound, gruu

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE

Content-Type: application/sdp

Content-Length: 531

Contact: <sip:SOFTPHONE_B at 128.171.168.233
;transport=udp>;expires=3599;+sip.instance="<urn:uuid:ddb06f51-0843-0074-81e5-73487c17342d>"

User-Agent: Linphone Desktop/4.2.2 (macOS 10.15, Qt 5.14.2)
LinphoneCore/4.4.0-13-gc99cb9c88


v=0

o=SOFTPHONE_B 1261 3707 IN IP4 128.171.168.233

s=Talk

c=IN IP4 128.171.168.233

t=0 0

a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics

m=audio 7078 RTP/AVP 96 97 98 0 8 18 101 99 100

a=rtpmap:96 opus/48000/2

a=fmtp:96 useinbandfec=1

a=rtpmap:97 speex/16000

a=fmtp:97 vbr=on

a=rtpmap:98 speex/8000

a=fmtp:98 vbr=on

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/48000

a=rtpmap:99 telephone-event/16000

a=rtpmap:100 telephone-event/8000

a=rtcp-fb:* trr-int 1000

a=rtcp-fb:* ccm tmmbr


<--- Transmitting SIP response (479 bytes) to UDP:128.171.168.233:5060
--->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 128.171.168.233:5060
;rport=5060;received=128.171.168.233;branch=z9hG4bK.D-YbrxKYs

Call-ID: ziUzVUxYw7

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: <sip:200 at 128.171.77.23>;tag=z9hG4bK.D-YbrxKYs

CSeq: 20 INVITE

WWW-Authenticate: Digest
realm="asterisk",nonce="1612573994/b1f976725d3cbb6b1fc9af5923a87ac7",opaque="50221ed627077186",algorithm=md5,qop="auth"

Server: Asterisk PBX 16.14.0

Content-Length:  0



<--- Received SIP request (412 bytes) from UDP:128.171.168.233:5060 --->

ACK sip:200 at 128.171.77.23 SIP/2.0

Via: SIP/2.0/UDP 128.171.168.233:5060;branch=z9hG4bK.D-YbrxKYs;rport

Call-ID: ziUzVUxYw7

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: <sip:200 at 128.171.77.23>;tag=z9hG4bK.D-YbrxKYs

Contact: <sip:SOFTPHONE_B at 128.171.168.233
;transport=udp>;expires=3599;+sip.instance="<urn:uuid:ddb06f51-0843-0074-81e5-73487c17342d>"

Max-Forwards: 70

CSeq: 20 ACK



<--- Received SIP request (1484 bytes) from UDP:128.171.168.233:5060 --->

INVITE sip:200 at 128.171.77.23 SIP/2.0

Via: SIP/2.0/UDP 128.171.168.233:5060;branch=z9hG4bK.HgO8RDlH4;rport

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: sip:200 at 128.171.77.23

CSeq: 21 INVITE

Call-ID: ziUzVUxYw7

Max-Forwards: 70

Supported: replaces, outbound, gruu

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE

Content-Type: application/sdp

Content-Length: 531

Contact: <sip:SOFTPHONE_B at 128.171.168.233
;transport=udp>;expires=3599;+sip.instance="<urn:uuid:ddb06f51-0843-0074-81e5-73487c17342d>"

User-Agent: Linphone Desktop/4.2.2 (macOS 10.15, Qt 5.14.2)
LinphoneCore/4.4.0-13-gc99cb9c88

Authorization:  Digest realm="asterisk",
nonce="1612573994/b1f976725d3cbb6b1fc9af5923a87ac7", algorithm=md5,
opaque="50221ed627077186", username="SOFTPHONE_B", 
uri="
sip:200 at 128.171.77.23",
response="352ca45cd5adc103f4b679713905bde9",
cnonce="7F142IC~o5UVxMll", nc=00000001, qop=auth


v=0

o=SOFTPHONE_B 1261 3707 IN IP4 128.171.168.233

s=Talk

c=IN IP4 128.171.168.233

t=0 0

a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics

m=audio 7078 RTP/AVP 96 97 98 0 8 18 101 99 100

a=rtpmap:96 opus/48000/2

a=fmtp:96 useinbandfec=1

a=rtpmap:97 speex/16000

a=fmtp:97 vbr=on

a=rtpmap:98 speex/8000

a=fmtp:98 vbr=on

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/48000

a=rtpmap:99 telephone-event/16000

a=rtpmap:100 telephone-event/8000

a=rtcp-fb:* trr-int 1000

a=rtcp-fb:* ccm tmmbr


  == Setting global variable 'SIPDOMAIN' to '128.171.77.23'

<--- Transmitting SIP response (305 bytes) to UDP:128.171.168.233:5060
--->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 128.171.168.233:5060
;rport=5060;received=128.171.168.233;branch=z9hG4bK.HgO8RDlH4

Call-ID: ziUzVUxYw7

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: <sip:200 at 128.171.77.23>

CSeq: 21 INVITE

Server: Asterisk PBX 16.14.0

Content-Length:  0


    -- Executing [200 at sets:1] Answer("PJSIP/SOFTPHONE_B-00000015",
"") in
new stack

       > 0x2a1ec80 -- Strict RTP learning after remote address set to:
128.171.168.233:7078

<--- Transmitting SIP response (797 bytes) to UDP:128.171.168.233:5060
--->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 128.171.168.233:5060
;rport=5060;received=128.171.168.233;branch=z9hG4bK.HgO8RDlH4

Call-ID: ziUzVUxYw7

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: <sip:200 at 128.171.77.23>;tag=9fac7f96-d0e8-474a-ae22-c034315670b4

CSeq: 21 INVITE

Server: Asterisk PBX 16.14.0

Contact: <sip:128.171.77.23:5060>

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:   227


v=0

o=- 1261 3709 IN IP4 128.171.77.23

s=Asterisk

c=IN IP4 128.171.77.23

t=0 0

m=audio 19864 RTP/AVP 0 100

a=rtpmap:0 PCMU/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-16

a=ptime:20

a=maxptime:150

a=sendrecv


<--- Received SIP request (676 bytes) from UDP:128.171.168.233:5060 --->

ACK sip:128.171.77.23:5060 SIP/2.0

Via: SIP/2.0/UDP 128.171.168.233:5060;rport;branch=z9hG4bK.63-kP~vZY

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: <sip:200 at 128.171.77.23>;tag=9fac7f96-d0e8-474a-ae22-c034315670b4

CSeq: 21 ACK

Call-ID: ziUzVUxYw7

Max-Forwards: 70

Authorization:  Digest realm="asterisk",
nonce="1612573994/b1f976725d3cbb6b1fc9af5923a87ac7", algorithm=md5,
opaque="50221ed627077186", username="SOFTPHONE_B", 
uri="
sip:200 at 128.171.77.23",
response="352ca45cd5adc103f4b679713905bde9",
cnonce="7F142IC~o5UVxMll", nc=00000001, qop=auth

User-Agent: Linphone Desktop/4.2.2 (macOS 10.15, Qt 5.14.2)
LinphoneCore/4.4.0-13-gc99cb9c88



    -- Executing [200 at sets:2]
Playback("PJSIP/SOFTPHONE_B-00000015", "
hello-world") in new stack

    -- <PJSIP/SOFTPHONE_B-00000015> Playing 'hello-world.slin'
(language
'en')

       > 0x2a1ec80 -- Strict RTP switching to RTP target address
128.171.168.233:7078 as source

    -- Executing [200 at sets:3] Hangup("PJSIP/SOFTPHONE_B-00000015",
"") in
new stack

  == Spawn extension (sets, 200, 3) exited non-zero on
'PJSIP/SOFTPHONE_B-00000015'

<--- Transmitting SIP request (432 bytes) to TLS:128.171.168.233:5061 --->

BYE sip:SOFTPHONE_B at 128.171.168.233;transport=udp SIP/2.0

Via: SIP/2.0/TLS 128.171.77.23:5061
;rport;branch=z9hG4bKPj41b05244-9271-43d8-8c2d-f28496b22179;alias

From: <sip:200 at 128.171.77.23>;tag=9fac7f96-d0e8-474a-ae22-c034315670b4

To: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

Call-ID: ziUzVUxYw7

CSeq: 6763 BYE

Reason: Q.850;cause=16

Max-Forwards: 70

User-Agent: Asterisk PBX 16.14.0

Content-Length:  0



<--- Received SIP request (677 bytes) from UDP:128.171.168.233:5060 --->

BYE sip:128.171.77.23:5060 SIP/2.0

Via: SIP/2.0/UDP 128.171.168.233:5060;branch=z9hG4bK.xNo4PqF4N;rport

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: <sip:200 at 128.171.77.23>;tag=9fac7f96-d0e8-474a-ae22-c034315670b4

CSeq: 22 BYE

Call-ID: ziUzVUxYw7

Max-Forwards: 70

User-Agent: Linphone Desktop/4.2.2 (macOS 10.15, Qt 5.14.2)
LinphoneCore/4.4.0-13-gc99cb9c88

Authorization:  Digest realm="asterisk",
nonce="1612573994/b1f976725d3cbb6b1fc9af5923a87ac7", algorithm=md5,
opaque="50221ed627077186", username="SOFTPHONE_B", 
uri="sip:
128.171.77.23:5060", response="1edae1a95308e6d2076a68099cfecb9a",
cnonce="5MRI3GsazLI35KUw", nc=00000002, qop=auth



<--- Transmitting SIP response (368 bytes) to UDP:128.171.168.233:5060
--->

SIP/2.0 481 Call/Transaction Does Not Exist

Via: SIP/2.0/UDP 128.171.168.233:5060
;rport=5060;received=128.171.168.233;branch=z9hG4bK.xNo4PqF4N

Call-ID: ziUzVUxYw7

From: "VOIP1_test" <sip:SOFTPHONE_B at
128.171.77.23>;tag=XvCbVpnIJ

To: <sip:200 at 128.171.77.23>;tag=9fac7f96-d0e8-474a-ae22-c034315670b4

CSeq: 22 BYE

Server: Asterisk PBX 16.14.0

Content-Length:  0


Here's what happens when using a udp hardphone (on ext 101) to call a tls
hardphone (on ext 100). It went straight to the no-body-around message w/o
ringing and on-hold music.

<--- Received SIP request (1095 bytes) from UDP:128.171.77.48:50906 --->

INVITE sip:100 at 128.171.77.23 SIP/2.0

Via: SIP/2.0/UDP 128.171.77.48:5060;branch=z9hG4bK1b7dab42

From: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
To: <sip:100 at 128.171.77.23>

Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

Max-Forwards: 70

Date: Sat, 06 Feb 2021 01:18:54 GMT

CSeq: 101 INVITE

User-Agent: Cisco-CP7940G/8.0

Contact: <sip:0000f30B0B02 at 128.171.77.48:5060;transport=udp>

Expires: 180

Accept: application/sdp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE

Remote-Party-ID: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub

Content-Length: 278

Content-Type: application/sdp

Content-Disposition: session;handling=optional


v=0

o=Cisco-SIPUA 25302 0 IN IP4 128.171.77.48

s=SIP Call

t=0 0

m=audio 25298 RTP/AVP 0 8 18 101

c=IN IP4 128.171.77.48

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv


<--- Transmitting SIP response (529 bytes) to UDP:128.171.77.48:5060 --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 128.171.77.48:5060
;received=128.171.77.48;branch=z9hG4bK1b7dab42

Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

From: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
To: <sip:100 at 128.171.77.23>;tag=z9hG4bK1b7dab42

CSeq: 101 INVITE

WWW-Authenticate: Digest
realm="asterisk",nonce="1612574334/f3c178b654f33965ce8a5aa89a50bcd0",opaque="654d62cc2c5bb89c",algorithm=md5,qop="auth"

Server: Asterisk PBX 16.14.0

Content-Length:  0



<--- Received SIP request (372 bytes) from UDP:128.171.77.48:52171 --->

ACK sip:100 at 128.171.77.23 SIP/2.0

Via: SIP/2.0/UDP 128.171.77.48:5060;branch=z9hG4bK1b7dab42

From: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
To: <sip:100 at 128.171.77.23>;tag=z9hG4bK1b7dab42

Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

Date: Sat, 06 Feb 2021 01:18:54 GMT

CSeq: 101 ACK

Content-Length: 0



<--- Received SIP request (1362 bytes) from UDP:128.171.77.48:50906 --->

INVITE sip:100 at 128.171.77.23 SIP/2.0

Via: SIP/2.0/UDP 128.171.77.48:5060;branch=z9hG4bK6781e064

From: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
To: <sip:100 at 128.171.77.23>

Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

Max-Forwards: 70

Date: Sat, 06 Feb 2021 01:18:54 GMT

CSeq: 102 INVITE

User-Agent: Cisco-CP7940G/8.0

Contact: <sip:0000f30B0B02 at 128.171.77.48:5060;transport=udp>

Authorization: Digest
username="0000f30B0B02",realm="asterisk",uri="
sip:100 at 128.171.77.23
",response="e4805ed40663b1345932a634488941b4",nonce="1612574334/f3c178b654f33965ce8a5aa89a50bcd0",opaque="654d62cc2c5bb89c",cnonce="28e5aea2",qop=auth,nc=00000001,algorithm=md5

Expires: 180

Accept: application/sdp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE

Remote-Party-ID: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub

Content-Length: 278

Content-Type: application/sdp

Content-Disposition: session;handling=optional


v=0

o=Cisco-SIPUA 25302 0 IN IP4 128.171.77.48

s=SIP Call

t=0 0

m=audio 25298 RTP/AVP 0 8 18 101

c=IN IP4 128.171.77.48

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv


  == Setting global variable 'SIPDOMAIN' to '128.171.77.23'

<--- Transmitting SIP response (357 bytes) to UDP:128.171.77.48:5060 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 128.171.77.48:5060
;received=128.171.77.48;branch=z9hG4bK6781e064

Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

From: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
To: <sip:100 at 128.171.77.23>

CSeq: 102 INVITE

Server: Asterisk PBX 16.14.0

Content-Length:  0



    -- Executing [100 at sets:1] Dial("PJSIP/0000f30B0B02-00000016",
"
PJSIP/0000f30A0A01,10,m") in new stack

    -- Called PJSIP/0000f30A0A01

    -- Started music on hold, class 'default', on channel
'PJSIP/0000f30B0B02-00000016'

       > 0x7f0fa80057f0 -- Strict RTP learning after remote address set to:
128.171.77.48:25298

<--- Transmitting SIP response (813 bytes) to UDP:128.171.77.48:5060 --->

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 128.171.77.48:5060
;received=128.171.77.48;branch=z9hG4bK6781e064

Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

From: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
To: <sip:100 at 128.171.77.23>;tag=b96a041f-35c1-406f-8cd5-7029dfaad3f4

CSeq: 102 INVITE

Server: Asterisk PBX 16.14.0

Contact: <sip:128.171.77.23:5060>

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE, REFER

Content-Type: application/sdp

Content-Length:   225


v=0

o=- 25302 2 IN IP4 128.171.77.23

s=Asterisk

c=IN IP4 128.171.77.23

t=0 0

m=audio 17122 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv


  == Everyone is busy/congested at this time (1:0/1/0)

    -- Stopped music on hold on PJSIP/0000f30B0B02-00000016

    -- Executing [100 at sets:2]
Playback("PJSIP/0000f30B0B02-00000016", "
vm-nobodyavail") in new stack

<--- Transmitting SIP response (847 bytes) to UDP:128.171.77.48:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 128.171.77.48:5060
;received=128.171.77.48;branch=z9hG4bK6781e064

Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

From: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
To: <sip:100 at 128.171.77.23>;tag=b96a041f-35c1-406f-8cd5-7029dfaad3f4

CSeq: 102 INVITE

Server: Asterisk PBX 16.14.0

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE, REFER

Contact: <sip:128.171.77.23:5060>

Supported: 100rel, timer, replaces, norefersub

Content-Type: application/sdp

Content-Length:   225


v=0

o=- 25302 2 IN IP4 128.171.77.23

s=Asterisk

c=IN IP4 128.171.77.23

t=0 0

m=audio 17122 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv


       > 0x7f0fa80057f0 -- Strict RTP switching to RTP target address
128.171.77.48:25298 as source

    -- <PJSIP/0000f30B0B02-00000016> Playing 'vm-nobodyavail.slin'
(language 'en')

<--- Received SIP request (834 bytes) from UDP:128.171.77.48:50906 --->

ACK sip:128.171.77.23:5060 SIP/2.0

Via: SIP/2.0/UDP 128.171.77.48:5060;branch=z9hG4bK309268b1

From: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
To: <sip:100 at 128.171.77.23>;tag=b96a041f-35c1-406f-8cd5-7029dfaad3f4

Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

Max-Forwards: 70

Date: Sat, 06 Feb 2021 01:18:55 GMT

CSeq: 102 ACK

User-Agent: Cisco-CP7940G/8.0

Authorization: Digest
username="0000f30B0B02",realm="asterisk",uri="
sip:100 at 128.171.77.23
",response="e4805ed40663b1345932a634488941b4",nonce="1612574334/f3c178b654f33965ce8a5aa89a50bcd0",opaque="654d62cc2c5bb89c",cnonce="28e5aea2",qop=auth,nc=00000001,algorithm=md5

Remote-Party-ID: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;party=calling;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0



    -- Executing [100 at sets:3] Hangup("PJSIP/0000f30B0B02-00000016",
"") in
new stack

  == Spawn extension (sets, 100, 3) exited non-zero on
'PJSIP/0000f30B0B02-00000016'

<--- Transmitting SIP request (499 bytes) to UDP:128.171.77.48:5060 --->

BYE sip:0000f30B0B02 at 128.171.77.48:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 128.171.77.23:5060
;rport;branch=z9hG4bKPje24627f6-0b3b-4b65-ab88-935a16a1100c

From: <sip:100 at 128.171.77.23>;tag=b96a041f-35c1-406f-8cd5-7029dfaad3f4

To: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

CSeq: 29223 BYE

Reason: Q.850;cause=34

Max-Forwards: 70

User-Agent: Asterisk PBX 16.14.0

Content-Length:  0



<--- Received SIP response (439 bytes) from UDP:128.171.77.48:50906 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 128.171.77.23:5060
;rport;branch=z9hG4bKPje24627f6-0b3b-4b65-ab88-935a16a1100c

From: <sip:100 at 128.171.77.23>;tag=b96a041f-35c1-406f-8cd5-7029dfaad3f4

To: "Ciscophone_325" <sip:0000f30B0B02 at
128.171.77.23>;tag=00075083381f5e4813be2318-77037fde
Call-ID: 00075083-381f0006-5e26f0de-139ba68a at 128.171.77.48

Date: Sat, 06 Feb 2021 01:18:58 GMT

CSeq: 29223 BYE

Server: Cisco-CP7940G/8.0

Content-Length: 0

  Thanks,

--Ruisheng

On Wed, Feb 3, 2021 at 11:44 PM Joshua C. Colp <jcolp at digium.com>
wrote:
> On Wed, Feb 3, 2021 at 11:02 PM Ruisheng Peng <rpeng at
ifa.hawaii.edu>
> wrote:
>
> <snip>
>
> When using handsets with udp or tcp transports to dial ext 100, it'd
>> hangup after the no-one-arround message.  However, when using the
handset
>> with tls transport, it doesn't hang up on its own if ext 100 is not
>> answered.  I have to click the hangup button to accomplish that. 
Here's
>> what asterisk log shows:
>>
>>   == Setting global variable 'SIPDOMAIN' to
'128.171.77.23'
>>
>>     -- Executing [100 at sets:1]
Dial("PJSIP/SOFTPHONE_B-00000007", "
>> PJSIP/0000f30A0A01,10,m") in new stack
>>
>>     -- Called PJSIP/0000f30A0A01
>>
>>     -- Started music on hold, class 'default', on channel
>> 'PJSIP/SOFTPHONE_B-00000007'
>>
>>        > 0x7f0fa801ede0 -- Strict RTP learning after remote address
set
>> to: 128.171.168.233:7078
>>
>>     -- PJSIP/0000f30A0A01-00000008 is ringing
>>
>>     -- PJSIP/0000f30A0A01-00000008 is ringing
>>
>>        > 0x7f0fa801ede0 -- Strict RTP switching to RTP target
address
>> 128.171.168.233:7078 as source
>>
>>        > 0x7f0fa801ede0 -- Strict RTP learning complete - Locking on
>> source address 128.171.168.233:7078
>>
>>     -- Nobody picked up in 10000 ms
>>
>>     -- Stopped music on hold on PJSIP/SOFTPHONE_B-00000007
>>
>>     -- Executing [100 at sets:2]
Playback("PJSIP/SOFTPHONE_B-00000007", "
>> vm-nobodyavail") in new stack
>>
>>     -- <PJSIP/SOFTPHONE_B-00000007> Playing
'vm-nobodyavail.slin'
>> (language 'en')
>>
>>     -- Executing [100 at sets:3]
Hangup("PJSIP/SOFTPHONE_B-00000007", "")
>> in new stack
>>
>>   == Spawn extension (sets, 100, 3) exited non-zero on
>> 'PJSIP/SOFTPHONE_B-00000007'
>> voip1*CLI>
>>
>>  Another quirk is when I use a phone with udp transport (RP_Yealink) to
>> call a phone with tls transport (RP_OMBP) it immediately jumps
>> the no-one-around message w/o ringing, then hang up.  The tls phone is
>> shown available but asterisk sees it busy:
>>
>>   == Setting global variable 'SIPDOMAIN' to
'128.171.77.23'
>>
>>     -- Executing [103 at sets:1]
Dial("PJSIP/0000f30A0A01-0000000d", "
>> PJSIP/SOFTPHONE_B,10") in new stack
>>
>>     -- Called PJSIP/SOFTPHONE_B
>>
>>   == Everyone is busy/congested at this time (1:0/1/0)
>>
>>     -- Executing [103 at sets:2]
Playback("PJSIP/0000f30A0A01-0000000d", "
>> vm-nobodyavail") in new stack
>>
>>        > 0x7f0fa000c330 -- Strict RTP learning after remote address
set
>> to: 128.171.77.118:11790
>>
>>        > 0x7f0fa000c330 -- Strict RTP switching to RTP target
address
>> 128.171.77.118:11790 as source
>>
>>     -- <PJSIP/0000f30A0A01-0000000d> Playing
'vm-nobodyavail.slin'
>> (language 'en')
>>
>>     -- Executing [103 at sets:3]
Hangup("PJSIP/0000f30A0A01-0000000d", "")
>> in new stack
>>
>>   == Spawn extension (sets, 103, 3) exited non-zero on
>> 'PJSIP/0000f30A0A01-0000000d'
>>
>> voip1*CLI>
>>
>>   Suppose it's not cool to mix transports among your handsets? Any
>> suggestions?
>>
>
> I'd suggest looking at the actual SIP signaling to see what is going on
> using "pjsip set logger on" and also providing configuration.
This would
> allow better insight into what exactly is going on.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Joshua C. Colp

2021-Feb-08 22:26 UTC

head link

[asterisk-users] Hangup() not working for handsets using pls transport?

On Mon, Feb 8, 2021 at 6:14 PM Ruisheng Peng <rpeng at ifa.hawaii.edu>
wrote:
> Thanks Jashua for the suggestion.  To find out if the issue was only
> limited to the softphone that was using tls transport (SOFTPHONE_B on ext
> 103, a linphone running off my MBP), I also turned one of the hard phone
> (0000f30A0A01 on ext 100, a Yealink T32G) into using tls transport.  It
> behaves similarly to the linphone in that the Hangup() call in dialplan is
> silently ignored, and the handsets would alway appear as busy/unavilable.
>
Have you configured the devices, on them or using their provisioning, to
use TLS? It does not appear so as they are using UDP, while you're forcing
a TLS transport in Asterisk. This would not work.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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asterisk users - Feb 2021 - Hangup() not working for handsets using pls transport?

[asterisk-users] Hangup() not working for handsets using pls transport?

[asterisk-users] Hangup() not working for handsets using pls transport?