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Joshua C. Colp

2021-Feb-04 09:43 UTC

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

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
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Ruisheng Peng

2021-Feb-06 01:29 UTC

head link

[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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> To UNSUBSCRIBE or update options visit:
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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?