[asterisk-users] Problem in dialing out

[asterisk asterisk]

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:secret at 202.0.179.3

[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:asterisk at sip.xxxxx.xxx>;tag=as2d14b9ec
To: <sip:202.0.179.3>;tag=6b0704d0
CSeq: 102 OPTIONS
Call-ID: 17e0315c21d7dbc10e8c185740e21148 at sip.xxxxx.xxx
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
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