[asterisk-users] Problem in dialing out

[Rizwan Hisham]

try this

http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

On Sat, Feb 19, 2011 at 5:00 AM, asterisk asterisk <asterisk at ck-lee.com>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: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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-- 
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 3333 6767 26
E: rizwanhasham at gmail.com
W: www.axvoice.com
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