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Ruisheng Peng

2021-Feb-04 02:11 UTC

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

Hi all,

  I managed to get tls transport going with asterisk 16.14.0, and set one
handset (SOFTPHONE_B) to use the transport.  I have set up a few other
handsets (both soft and hard) to use udp and tcp transports:

voip1*CLI> pjsip show endpoints


 Endpoint:  <Endpoint/CID.....................................>
<State.....>  <Channels.>

    I/OAuth:
<AuthId/UserName...........................................................>

        Aor:  <Aor............................................>
<MaxContact>

      Contact:  <Aor/ContactUri..........................>
<Hash....>
<Status> <RTT(ms)..>

  Transport:  <TransportId........>  <Type>  <cos> 
<tos>
<BindAddress..................>

   Identify:
<Identify/Endpoint.........................................................>

        Match:  <criteria.........................>

    Channel:  <ChannelId......................................>
<State.....>  <Time.....>

        Exten: <DialedExten...........>  CLCID:
<ConnectedLineCID.......>

=========================================================================================

 Endpoint:  0000f30A0A01                                         Not in
use    0 of inf

     InAuth:  0000f30A0A01/0000f30A0A01

        Aor:  0000f30A0A01                                       1

      Contact:  0000f30A0A01/sip:0000f30A0A01 at 128.171.77.1 4800418965
NonQual         nan

  Transport:  transport-udp             udp      0      0  0.0.0.0:5060


 Endpoint:  0000f30B0B02                                         Not in
use    0 of inf

     InAuth:  0000f30B0B02/0000f30B0B02

        Aor:  0000f30B0B02                                       1

      Contact:  0000f30B0B02/sip:0000f30B0B02 at 128.171.77.4 615cc2a2c6
NonQual         nan

  Transport:  transport-udp             udp      0      0  0.0.0.0:5060


 Endpoint:  SOFTPHONE_A
Unavailable   0 of inf

     InAuth:  SOFTPHONE_A/SOFTPHONE_A

        Aor:  SOFTPHONE_A                                        2

  Transport:  transport-tcp             tcp      0      0  0.0.0.0:5060


 Endpoint:  SOFTPHONE_B                                          Not in
use    0 of inf

     InAuth:  SOFTPHONE_B/SOFTPHONE_B

        Aor:  SOFTPHONE_B                                        2

      Contact:  SOFTPHONE_B/sip:SOFTPHONE_B at 128.171.168.23 78257ab30a
NonQual         nan

  Transport:  transport-tls             tls      0      0  0.0.0.0:5061



Objects found: 4


voip1*CLI>

For testing, I have the following in /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()


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?

  Thanks,

--Ruisheng
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Joshua C. Colp

2021-Feb-04 09:43 UTC

head link

[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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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?