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

[Joshua C. Colp]

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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