try this http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
On Sat, Feb 19, 2011 at 5:00 AM, asterisk asterisk <[email protected]>wrote: > I have a sip trunk connecting to a huawei softx3000. At the moment, I can > register and dial in. > > However, peer status shows not reachable > > sip show peer as follow > > * Name : cmphone > Secret : <Set> > MD5Secret : <Not set> > Remote Secret: <Not set> > Context : from-cmphone > Subscr.Cont. : device-hints > Language : > AMA flags : Unknown > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup : > Pickupgroup : > MOH Suggest : > Mailbox : > VM Extension : asterisk > LastMsgsSent : 32767/65535 > Call limit : 0 > Max forwards : 0 > Dynamic : No > Callerid : "" <> > MaxCallBR : 384 kbps > Expire : -1 > 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 : No > Subscriptions: Yes > Overlap dial : Yes > Outb. proxy : 202.0.179.3 > DTMFmode : rfc2833 > Timer T1 : 500 > Timer B : 32000 > ToHost : 202.0.179.3 > Addr->IP : 202.0.179.3:5060 > Defaddr->IP : (null) > Prim.Transp. : UDP > Allowed.Trsp : UDP > Def. Username: 852350xxxxxx > SIP Options : 100rel > Codecs : 0xe (gsm|ulaw|alaw) > Codec Order : (alaw:20,ulaw:20,gsm:20) > Auto-Framing : No > 100 on REG : No > Status : UNREACHABLE > Useragent : > Reg. Contact : > Qualify Freq : 60000 ms > Sess-Timers : Accept > Sess-Refresh : uas > Sess-Expires : 1800 secs > Min-Sess : 90 secs > RTP Engine : asterisk > Parkinglot : > Use Reason : No > Encryption : No > > In sip.conf > > I have > > register = 852350xxxxx:[email protected] > > [cmphone] > type = friend > host = 202.0.179.3 > secret = secret > username = 852350xxxxx > context = from-cmphone > dtmfmode = rfc2833 > outboundproxy = 202.0.179.3 > caninvite=no > insecure = port,invite > nat = yes > > When debug is on, the error message is > > > <--- SIP read from UDP:202.0.179.3:5060 ---> > SIP/2.0 504 Server Time-out > From: "asterisk" <sip:[email protected]>;tag=as2d14b9ec > To: <sip:202.0.179.3>;tag=6b0704d0 > CSeq: 102 OPTIONS > Call-ID: [email protected] > Via: SIP/2.0/UDP > 14.xxx.xxx.xxx:5060;branch=z9hG4bK3646eaf2;received=14.xxx.xxx.xxx;rport=5060 > Content-Length: 0 > > Any help is appreciate. > > CK > > -- > _____________________________________________________________________ > -- 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 > -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: [email protected] W: www.axvoice.com
-- _____________________________________________________________________ -- 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
