try this<br><br><a href="http://www.voip-info.org/wiki/view/Asterisk+sip+qualify">http://www.voip-info.org/wiki/view/Asterisk+sip+qualify</a><br><br><div class="gmail_quote">On Sat, Feb 19, 2011 at 5:00 AM, asterisk asterisk <span dir="ltr"><<a href="mailto:asterisk@ck-lee.com">asterisk@ck-lee.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I have a sip trunk connecting to a huawei softx3000. At the moment, I can register and dial in.<br>
<br>However, peer status shows not reachable<br><br>sip show peer as follow<br><br> * Name : cmphone<br> Secret : <Set><br>
MD5Secret : <Not set><br> Remote Secret: <Not set><br> Context : from-cmphone<br> Subscr.Cont. : device-hints<br> Language :<br> AMA flags : Unknown<br> Transfer mode: open<br> CallingPres : Presentation Allowed, Not Screened<br>
Callgroup :<br> Pickupgroup :<br> MOH Suggest :<br> Mailbox :<br> VM Extension : asterisk<br> LastMsgsSent : 32767/65535<br> Call limit : 0<br> Max forwards : 0<br> Dynamic : No<br> Callerid : "" <><br>
MaxCallBR : 384 kbps<br> Expire : -1<br> Insecure : port,invite<br> Force rport : Yes<br> ACL : No<br> DirectMedACL : No<br> T.38 support : No<br> T.38 EC mode : Unknown<br> T.38 MaxDtgrm: -1<br>
DirectMedia : Yes<br> PromiscRedir : No<br> User=Phone : No<br> Video Support: No<br> Text Support : No<br> Ign SDP ver : No<br> Trust RPID : No<br> Send RPID : No<br> Subscriptions: Yes<br> Overlap dial : Yes<br>
Outb. proxy : 202.0.179.3<br> DTMFmode : rfc2833<br> Timer T1 : 500<br> Timer B : 32000<br> ToHost : 202.0.179.3<br> Addr->IP : <a href="http://202.0.179.3:5060" target="_blank">202.0.179.3:5060</a><br>
Defaddr->IP : (null)<br>
Prim.Transp. : UDP<br> Allowed.Trsp : UDP<br> Def. Username: 852350xxxxxx<br> SIP Options : 100rel<br> Codecs : 0xe (gsm|ulaw|alaw)<br> Codec Order : (alaw:20,ulaw:20,gsm:20)<br> Auto-Framing : No<br> 100 on REG : No<br>
Status : UNREACHABLE<br> Useragent :<br> Reg. Contact :<br> Qualify Freq : 60000 ms<br> Sess-Timers : Accept<br> Sess-Refresh : uas<br> Sess-Expires : 1800 secs<br> Min-Sess : 90 secs<br> RTP Engine : asterisk<br>
Parkinglot :<br> Use Reason : No<br> Encryption : No<br><br>In sip.conf<br><br>I have<br><br>register = <a href="mailto:852350xxxxx%3Asecret@202.0.179.3" target="_blank">852350xxxxx:secret@202.0.179.3</a><br><br>
[cmphone]<br>type = friend<br>
host = 202.0.179.3<br>secret = secret<br>username = 852350xxxxx<br>context = from-cmphone<br>dtmfmode = rfc2833<br>outboundproxy = 202.0.179.3<br>caninvite=no<br>insecure = port,invite<br>nat = yes<br><br>When debug is on, the error message is<br>
<br><br><--- SIP read from UDP:<a href="http://202.0.179.3:5060" target="_blank">202.0.179.3:5060</a> ---><br>SIP/2.0 504 Server Time-out<br>From: "asterisk" <sip:asterisk@sip.xxxxx.xxx>;tag=as2d14b9ec<br>
To: <sip:202.0.179.3>;tag=6b0704d0<br>
CSeq: 102 OPTIONS<br>Call-ID: 17e0315c21d7dbc10e8c185740e21148@sip.xxxxx.xxx<br>Via: SIP/2.0/UDP 14.xxx.xxx.xxx:5060;branch=z9hG4bK3646eaf2;received=14.xxx.xxx.xxx;rport=5060<br>Content-Length: 0<br><br>Any help is appreciate.<br>
<br>CK<br>
<br>--<br>
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