asterisk users - Aug 2020 - Channels freeze on Confbridge

    I am having a strange problem.  We have an Asterisk 16.12.0 server 
(we have upgraded at least two versions since we found the problem) 
where users complain that confbridge calls end after about 30 minutes or 
so.  The problem is that according to Asterisk the calls are still 
active.  All users are cut off at the same time but a "core show 
channels verbose" still shows channels as active:

CBAnn/902-0000002f;1 default              s                   1 Up      
(None)       (Empty) 04:03:43
CBAnn/902-0000002f;2 default              s                   1 Up      
(None)       (Empty) 04:03:43                         6e7710ea-7c0f-4c7e-a

CBAnn/903-00000036;2 default              s                   1 Up      
(None)       (Empty) 02:47:04                         05e10e42-85ec-4120-b
CBAnn/903-00000036;1 default              s                   1 Up      
(None)       (Empty) 02:47:04

PJSIP/directo-0001b7 oficina              903                 2 Up      
ConfBridge   903                       8110221265 02:40:43 general     
general     05e10e42-85ec-4120-b

PJSIP/directo-0001af oficina              902                 2 Up      
ConfBridge   902                       8992596823 04:25:50 general     
general     6e7710ea-7c0f-4c7e-a

I have to manually hangup the channels.  The PSTN provider is a SIP trunk.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161

On 2020-08-18 13:00, Carlos Chavez wrote:
> users complain that confbridge calls end after about 30 minutes or so
You might want to turn up SIP debug logging -- could be a re-INVITE is getting
dropped, NAT pin-hole is closing, or some other network issue.

-- 
🤠 C. Maj, Technology Captain @ Penguin PBX Solutions
📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729)
🤙 International & SMS Texting +1.720.32.42.72.9
🐧 Visit on the World Wide Web at PENGUINPBX.COM

I had a similiar problem, but with calls dropping after 30 sec.
It turned out that Android didn't support RP-CID (Reverse Party Caller ID)
so when I sent the name of the callee to the caller (as some sort of
"centralized phonebook function") it caused calls to be dropped as
android refused to reply on the packets or sent rejections back.

Check if you have some equipment on the line which doesn't support a
specific function, and configure the equipment to use a separate SIP account
with these features turned off.

I first tought it would just ignore unsupported features, but it turned out it
outright rejects packets with unsupported features.

Best regards, Sebastian Nielsen

-----Ursprungligt meddelande-----
Från: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at
lists.digium.com> För C.Maj
Skickat: den 22 augusti 2020 20:03
Till: asterisk-users at lists.digium.com
Ämne: Re: [asterisk-users] Channels freeze on Confbridge

On 2020-08-18 13:00, Carlos Chavez wrote:
> users complain that confbridge calls end after about 30 minutes or so
You might want to turn up SIP debug logging -- could be a re-INVITE is getting
dropped, NAT pin-hole is closing, or some other network issue.

-- 
🤠 C. Maj, Technology Captain @ Penguin PBX Solutions
📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729)
🤙 International & SMS Texting +1.720.32.42.72.9
🐧 Visit on the World Wide Web at PENGUINPBX.COM

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