Thank you Gareth I will try that :)
On 16 January 2014 14:55, Gareth Blades <mailinglist+aster...@dns99.co.uk>wrote: > 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 $TRACEFILE -C 10 -W > 500 & > > > > On 16/01/14 14:27, Tiago Geada wrote: > > You're right, seems like a nice way to debug. Regarding that, how would > the impact be affected running it on asterisk box? I guess only port 5060 > is not too bad > > > On 16 January 2014 14:09, Gareth Blades > <mailinglist+aster...@dns99.co.uk>wrote: > >> On 16/01/14 10:47, Tiago Geada wrote: >> >> Hi folks, >> >> We've been having a weird issue... It is happening more often in the >> last few months... >> >> Most inbound calls, we have in our dialplan before Queue(): >> >> Set(CALLERID(name)=${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 >> > > > > > -- > _____________________________________________________________________ > -- 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 >
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