Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache.
For example * Name : 501 Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pack-local Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : 501@local VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 5 Max forwards : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 3326 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : x.x.x.x:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: PACK501 SIP Options : (none) Codecs : 0x10c (ulaw|alaw|g729) Codec Order : (g729:20,alaw:20,ulaw:20) Auto-Framing : No 100 on REG : Yes Status : OK (17 ms) Useragent : snom870/8.4.20 Reg. Contact : sip:[email protected]:52753;line=hu9fedy3 Qualify Freq : 120000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No But in the DB I have clearly set nat to yes select name,nat from sip where name ='501'; +------+-----+ | name | nat | +------+-----+ | 501 | yes | +------+-----+ In all previous versions of asterisk we have used with realtime we would see a line in the sip show peer looking like: Nat : Always Has the table definition changed in asterisk 1.8.3? Is there a bug stopping this value being picked up? Can someone even point me to the correct source files so I can attempt to try and work out the correct 1.8 sip table definition from there as I can't find one anywhere at all? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _____________________________________________________________________ -- 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
