[asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Gareth Blades]

Very little as the amount of data being captured is quite small. We have 
it running on our production servers which routinely handle a couple of 
hundred concurrent calls.

This is the script we use to start off the capture. It uses rolling 
capture files so we will always have the last X number of capture logs. 
It works very well and we have a custom system which enables us to 
search for calls and request traces for them for when we have to 
diagnose problems.

#!/bin/bash
cd /var/lib/asterisk/siptraces
DATE=`date +%Y%m%d%H%M%S`
TRACEFILE=/var/lib/asterisk/siptraces/$DATE-siptrace.pcap
nohup /usr/sbin/tcpdump -p -i eth0 -s 0 port 5060 -w $TRACEFILE -C 10 -W 
500 &


On 16/01/14 14:27, Tiago Geada wrote:
> You're right, seems like a nice way to debug. Regarding that, how 
> would the impact be affected running it on asterisk box? I guess only 
> port 5060 is not too bad
>
>
> On 16 January 2014 14:09, Gareth Blades 
> <mailinglist+asterisk at dns99.co.uk 
> <mailto:mailinglist+asterisk at dns99.co.uk>> wrote:
>
>     On 16/01/14 10:47, Tiago Geada wrote:
>>     Hi folks,
>>
>>     We've been having a weird issue... It is happening more often in
>>     the last few months...
>>
>>     Most inbound calls, we have in our dialplan before Queue():
>>
>>     Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});
>>
>>     So when the call rings a member, softphone will show this string ....
>>
>>     The issue is that sometimes the string showing in the softphone
>>     is not the same. Its a string from a past call, in the latest
>>     case I've seen, from about 40 days ago!!
>>
>>     User took a screenshot, I've searched for that uniqueid showing
>>     in softphone in cdr, and that string was valid for a different
>>     call 40 days ago!!
>>
>>
>>     I searched full log, and Set() sets the correct string... I can't
>>     figure why softphone shows a string from a past call !!
>>
>>     :(
>>
>>     Any hints ?
>>
>>
>     I would leave tcpdump running capturing port 5060 so you can load
>     it onto wireshark and have a look at the sip headers. That will
>     tell you if the SIP is incorrect or if its a problem with the client.
>
>     --
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>     New to Asterisk? Join us for a live introductory webinar every Thurs:
>     http://www.asterisk.org/hello
>
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140116/c2b3dd8b/attachment.html>