> You mean "sip set debug on" ? Yes, that's correct for chan_sip. Sorry, I was vague -- there is now a different command for chan_pjsip, didn't know which you were using.
On Thu, May 28, 2015 at 12:49 PM, Ethy H. Brito <[email protected]> wrote: > On Thu, 28 May 2015 11:15:45 -0500 > Scott Griepentrog <[email protected]> wrote: > > > The string "5a2600300339934f704528bb14ed05e9@MyAsterisk:5060" is the > unique > > identifier for the call in SIP known as the Call-ID. If you have a > packet > > capture of the port 5060 SIP traffic, that identifier will be in each SIP > > message related to the call, which also includes the full from and to > > details. > > That is the problem. Since the message occurs typically about 2~3 times a > day (or even less), I will have tons of packets to sniff. > > But, I will give it a try. > > > > > As an alternative to running a separate packet capture, you can enable > SIP > > message logging in Asterisk, which puts the full SIP message into the > same > > log file. > > You mean "sip set debug on" ? > > > Be aware however that this can fill your hard drive quite > > rapidly, as well as put additional load on the disk storage system. > > I am pretty aware of that. Learn it the hard way. > > Cheers > > Ethy > > > > > > On Thu, May 28, 2015 at 11:03 AM, Ethy H. Brito <[email protected] > > > > wrote: > > > > > > > > Hi All > > > > > > I have a few lines like this at asterisk/messages. > > > > > > [May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call > > > 5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 - no reply to our > > > critical > > > packet (see > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > > ). > > > > > > Since we have hundreds of clients with hundreds of simultaneous calls, > how > > > is > > > it possible to know to which customer/IP those calls refer to? > > > > > > The above literature don't say much to help to narrow down the problem > > > scope. > > > > > > Cheers > > > > > > Ethy > > > > > > -- > > > _____________________________________________________________________ > > > -- 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 > > > > > > > > > > > -- > > [image: Digium logo] > > Scott Griepentrog > > Digium, Inc · Software Developer > > 445 Jan Davis Drive NW · Huntsville, AL 35806 · US > > direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 > > Check us out at: http://digium.com · http://asterisk.org > > > -- > > Ethy H. Brito /"\ > InterNexo Ltda. \ / CAMPANHA DA FITA ASCII - CONTRA MAIL HTML > +55 (12) 3797-6860 X ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL > S.J.Campos - Brasil / \ > > PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc > > -- > _____________________________________________________________________ > -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org
-- _____________________________________________________________________ -- 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
