Re: [asterisk-users] Channels freeze on Confbridge

[Carlos Chavez]

On 25/08/20 7:20, Andrew Yager wrote:

On Sun, 23 Aug 2020 at 18:23, Antony Stone 
> wrote:


On Saturday 22 August 2020 at 22:51:48, Sebastian Nielsen wrote:

> 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.

I can see the point you're making here, but what's going to do
this after 30
*minutes* of normal call?


Have seen plenty of ALGs do weird things like this. 30 minutes is a 
nice number, and nice enough that I'd go hunting for ALG issues. It's 
a good multiple of 3 minutes, and quite possibly is some big number 
someone thought to set in something that "no one would ever hit".


A tcpdump would probably show you what's going on if the logs are 
otherwise unclear, and you could also make sure you have sensible RTP 
timeout rules.


Andrew


    We are zeroing in on something with the SIP trunk provider.  We 
have are testing a new carrier and so far we have not seen the same problem.


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Re: [asterisk-users] Channels freeze on Confbridge

[Andrew Yager]

On Sun, 23 Aug 2020 at 18:23, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Saturday 22 August 2020 at 22:51:48, Sebastian Nielsen wrote:
>
> > 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.
>
> I can see the point you're making here, but what's going to do this after
> 30
> *minutes* of normal call?
>

Have seen plenty of ALGs do weird things like this. 30 minutes is a nice
number, and nice enough that I'd go hunting for ALG issues. It's a good
multiple of 3 minutes, and quite possibly is some big number someone
thought to set in something that "no one would ever hit".

A tcpdump would probably show you what's going on if the logs are otherwise
unclear, and you could also make sure you have sensible RTP timeout rules.

Andrew
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Re: [asterisk-users] Channels freeze on Confbridge

[Sebastian Nielsen]

>>I can see the point you're making here, but what's going to do this after 30
*minutes* of normal call?

I was more into, if there is some feature that somehow triggers after 30 
minutes of call - and this feature is unsupported on some client, which causes 
it to drop the call. For example, if you are trying to send some call cost 
notification for long calls out of band or similiar, and some devices doesn't 
support this feature.
 




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Re: [asterisk-users] Channels freeze on Confbridge

[Antony Stone]

On Saturday 22 August 2020 at 22:51:48, Sebastian Nielsen wrote:

> 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.

I can see the point you're making here, but what's going to do this after 30 
*minutes* of normal call?

> 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.


Antony.

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 please *don't* CC me.

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Re: [asterisk-users] Channels freeze on Confbridge

[Sebastian Nielsen]

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-boun...@lists.digium.com 
 För C.Maj
Skickat: den 22 augusti 2020 20:03
Till: asterisk-users@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.

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Re: [asterisk-users] Channels freeze on Confbridge

[C.Maj]

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.

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📞 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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[asterisk-users] Channels freeze on Confbridge

[Carlos Chavez]

    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/902-002f;1 default  s   1 Up  
(None)   (Empty) 04:03:43
CBAnn/902-002f;2 default  s   1 Up  
(None)   (Empty) 04:03:43 6e7710ea-7c0f-4c7e-a


CBAnn/903-0036;2 default  s   1 Up  
(None)   (Empty) 02:47:04 05e10e42-85ec-4120-b
CBAnn/903-0036;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.

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