asterisk users - Oct 2006 - duplicate "ghost" calls with long duration

Hello everybody,

I am running 1.2.10-BRIstuffed-0.3.0-PRE-1s with florz-patches on Linux 
2.6.16 with 4 HFC-Cards in TE-mode connected to 4 PtP 
ISDN-"Anlagenanschluesse". There are about 40 SIP-clients connected 
(mostly Sipura/Linksys PAP2, and some SNOMs and softclients) to this server.

Everything works fine, except that my CDR reports some very long 
_concurrent_ calls from one sip client to (an expensive) pstn 
destination. The CDR from my telco tells the same!

First case:

Sep. 05 2006 11:46:40 20 secs call from X to 0900xxx (valid call, 
micropayment)
Sep. 05 2006 11:52:18 3573 secs call from same X to 0900xxx (????)
Sep. 05 2006 11:53:24 3466 secs call from same X to 0900xxx (????)

X is a Sipura-connected hardphone. The guy swears he has only done the 
first call. Strangely the others are concurrent and to the same 
micropayment number, which itself disconnects callers after the 
micropayment value has been reached!

Second case:

Sep. 21 2006 14:30:22 35981(!) secs call from Y to 01805xxx (????)
Sep. 21 2006 14:31:00 1823 secs call from Z to 01805xxx (valid, 
conference provider call)
Sep. 21 2006 14:36:09 35634(!) secs call from Y to 01805xxx (????)

You see the valid call is made from a different caller, Z instead of Y. 
Y is also a sipura phone as well as Z. Y did never call this number itself.

Since my telco's (Deutsche Telekom) CDR tells me the same, this problem 
seems to be an asterisk internal one.

Any hints?

Regards,
Bjoern

Bjoern Metzdorf wrote:
>
> Everything works fine, except that my CDR reports some very long 
> _concurrent_ calls from one sip client to (an expensive) pstn 
> destination. The CDR from my telco tells the same!
> Any hints?
I run into that from time to time for this business account we have 
where channels were staying open for a long time so I made a script run 
from cron to hang up any extension over X amount of time:

/usr/sbin/asterisk -rx "show channels concise" |awk -F : '($11
> 5400)
{print "/usr/sbin/asterisk -rx \"soft hangup " $1
"\""} '|sh

This looks at any calls over 90 minutes then hangs it up. You can modify 
it for your issue say something like:

/usr/sbin/asterisk -rx "show channels concise" |awk -F : 
'/YOUR_X_SIPURA_NUMBER/'|awk -F : '($11 > 5400) {print 
"/usr/sbin/asterisk -rx \"soft hangup " $1 "\""}
'|sh

Not practical though for saving money... If someone is on for say 1 
minute and there is an issue with the channel not hanging up, 5399 
minutes would still be billed.

-- 
===================================================J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
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you can also try using

busydetect=yes

busycount=4

in your zapata.conf

Hopefuly you won't start getting sudden hang ups, due to false positives and
it will be helpful enough.

Alyed  

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On 2006-10-17 09:00:51 -0700, Bjoern Metzdorf 
 said:
>> I run into that from time to time for this business account we have 
>> where channels were staying open for a long time so I made a script run
>> from cron to hang up any extension over X amount of time:
>> 
>> /usr/sbin/asterisk -rx "show channels concise" |awk -F :
'($11 > 5400)
>> {print "/usr/sbin/asterisk -rx \"soft hangup " $1
"\""} '|sh
>> 
>> This looks at any calls over 90 minutes then hangs it up. You can 
>> modify it for your issue say something like:
>> 
>> /usr/sbin/asterisk -rx "show channels concise" |awk -F : 
>> '/YOUR_X_SIPURA_NUMBER/'|awk -F : '($11 > 5400) {print 
>> "/usr/sbin/asterisk -rx \"soft hangup " $1
"\""} '|sh
>> 
>> Not practical though for saving money... If someone is on for say 1 
>> minute and there is an issue with the channel not hanging up, 5399 
>> minutes would still be billed.
> 
> What version are you using?
> 
> I never had these issues with asterisk 1.0.x in 15 months. That leads 
> me to a problematic 1.2.x or to faulty bristuff-patches.
> 
> I will upgrade asterisk asap to latest 1.2.x and add an absolute 
> timeout to those destinations.
> 
> But: Are we the only ones experiencing this?
That really doesn't sound at all the same to me as what he is 
describing?  In his case (ie not hung up calls) if you are using SIP 
handsets, then the rtptimeout setting can cut the calls off when there 
is no audio data flowing.

Good Luck,
Marty

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