Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: "xxxxxxxxxx" <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as23a29r59To: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060>Contact: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>Call-ID: 2f17b2103ea4792d571e2dce7e14bb05 at xxx.xxx.xxx.xxxCSeq: 102 INVITEUser-Agent: Asterisk PBX 1.6.2.9Date: Tue, 26 Feb 2013 04:54:29 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Type: application/sdpContent-Length: 444 Thanks,Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130226/4a3df602/attachment.htm>
qasimakhan at gmail.com
2013-Feb-26 07:30 UTC
[asterisk-users] set time zone in sip debug logs
Hi Kamlesh, Asterisk give you very less control over SIP messaging. You can how ever add/remove/modify SIP headers from initial invite only. To modify a sip header you can use asterisk function "*SIP_HEADER(<name>)*". If you want to permanently change date why not change system date/time? Regards, -Qasim On Tue, Feb 26, 2013 at 11:13 AM, Kamlesh Kumar <kamlesh_kmr at hotmail.com>wrote:> Hello, > > Please suggest the way to change the time zone in below sip debug logs. > > INVITE sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060 SIP/2.0 > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rport > Max-Forwards: 70 > From: "xxxxxxxxxx" <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as23a29r59 > To: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060> > Contact: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx> > Call-ID: 2f17b2103ea4792d571e2dce7e14bb05 at xxx.xxx.xxx.xxx > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.2.9 > *Date: Tue, 26 Feb 2013 04:54:29 GMT* > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 444 > > Thanks, > Kamlesh > > -- > _____________________________________________________________________ > -- 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/20130226/d3c63e61/attachment.htm>