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
