<div dir="ltr"><div><br></div>So "SIP/2.0 403 Forbidden" is a valid response for "qualify purpose"<br><div class="gmail_extra">Thanks Brian!!<br></div><div class="gmail_extra">rv<br></div><div class="gmail_extra">
<br><br><div class="gmail_quote">2014-07-03 5:18 GMT-04:00 Brian LaVallee <span dir="ltr"><<a href="mailto:b.lavallee@globaltank.jp" target="_blank">b.lavallee@globaltank.jp</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Rafael,<br>
<br>
It's nothing to worry about -and- you might not be able to fix it. But<br>
it's nothing to worry about.<br>
<br>
--<br>
<br>
Asterisk is using OPTIONS like a ping, qualify=yes. Since 403 is a<br>
*valid* SIP reply, the remote SIP service is considered reachable.<br>
<br>
My carrier replies with "405 Method Not Allowed", but it still indicates<br>
the SIP connection is up and working.<br>
<br>
--<br>
<br>
Some carriers do not support OPTIONS. This is normally due to a proxy<br>
or other security mechanisms.<br>
<br>
Remember, OPTIONS is a request for what commands will be accepted.<br>
Sometime, you just don't want to advertise that kind of information.<br>
<br>
--<br>
<br>
Check an INBOUND call (INVITE) and it will typically show what the<br>
carrier "allows". If OPTIONS is not listed, there's nothing you can do.<br>
<br>
<br>
IP CARRIER_IP.sip > LOCAL_IP.sip: UDP, length 870<br>
E.....@.9.9:=...j.p".....n$BINVITE sip:<a href="tel:2125551111" value="+12125551111">2125551111</a>@LOCAL_IP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP<br>
CARRIER_IP:5060;branch=z9hG4bKdac2492a2a1a086867cfb73fb2b5c8ac<br>
Via: SIP/2.0/UDP PROXY_IP:5060;branch=z9hG4bK09B55db052ffec696bd<br>
From: <sip:2125559999@PROXY_IP:5060>;tag=gK094dc1e4<br>
To: <sip:<a href="tel:2125551111" value="+12125551111">2125551111</a>@CARRIER_IP:5060>;tag=as2953dd14<br>
Call-ID: 1980326667_35899190@PROXY_IP<br>
CSeq: 7852 INVITE<br>
Max-Forwards: 69<br>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,PRACK,UPDATE<br>
<snip><br>
Accept: application/sdp<br>
<br>
<br>
Sincerely,<br>
Brian LaVallee<br>
<div class="HOEnZb"><div class="h5"><br>
<br>
<br>
On 6/25/14, 11:30 PM, Rafael Visser wrote:<br>
> Hi gurus!!!<br>
><br>
> I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn<br>
> Every minute asterisk sends an OPTION Request, i beleived that it's related<br>
> to qualify functions.<br>
> The every minute annoyng answer of the pstn is "403 Forbidden".<br>
> Some people told that asterisk is not sending the username in the OPTION,<br>
> required by the pstn.<br>
><br>
><br>
> Taking a look of the example of rfc3261.txt (pg 67), we found "carol", so<br>
> it makingme see that i am missing some config.<br>
>>><br>
> OPTIONS <a href="mailto:sip%3Acarol@chicago.com">sip:carol@chicago.com</a> SIP/2.0<br>
> Via: SIP/2.0/UDP <a href="http://pc33.atlanta.com" target="_blank">pc33.atlanta.com</a>;branch=z9hG4bKhjhs8ass877<br>
> Max-Forwards: 70<br>
> To: <<a href="mailto:sip%3Acarol@chicago.com">sip:carol@chicago.com</a>><br>
> <<<br>
><br>
><br>
> Is it wright?<br>
> How can i instruct FREEPBX to send the username in the option request?<br>
><br>
> Sorry for this silly question but a found no answer googling.<br>
><br>
><br>
><br>
> Thans in advance.<br>
> rv<br>
><br>
><br>
><br>
> This is the debug of the case<br>
><br>
><br>
> Reliably Transmitting (NAT) to 201.217.31.XX:5060:<br>
> OPTIONS sip:<a href="tel:201.217.31.10" value="+12012173110">201.217.31.10</a> SIP/2.0<br>
> Via: SIP/2.0/UDP 18x.16.204.XXX:6060;branch=z9hG4bK1d8715df;rport<br>
> Max-Forwards: 70<br>
> From: "Unknown" <sip:59X212376XXX@186.16.204.XXX:6060>;tag=as4491c6af<br>
> To: <sip:<a href="tel:201.217.31.10" value="+12012173110">201.217.31.10</a>><br>
> Contact: <sip:59X212376XXX@18x.16.204.XXX:6060><br>
> Call-ID: 4f02699e2632410c359e1ee43a021dc7@186.16.204.XXX:6060<br>
> CSeq: 102 OPTIONS<br>
> User-Agent: FPBX-2.11.0(1.8.25.0)<br>
> Date: Wed, 25 Jun 2014 13:47:19 GMT<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,<br>
> PUBLISH<br>
> Supported: replaces, timer<br>
> Content-Length: 0<br>
><br>
><br>
> <--- SIP read from UDP:201.217.31.XX:5060 ---><br>
> SIP/2.0 403 Forbidden<br>
> Via: SIP/2.0/UDP<br>
> 18x.16.204.XXX:6060;received=18x.16.204.XXX;branch=z9hG4bK1d8715df;rport=5060<br>
> From: "Unknown" <sip:59X212376XXX@18x.16.204.XXX:6060>;tag=as4491c6af<br>
> To: <sip:201.217.31.XX>;tag=aprqngfrt-nm50ea10000c6<br>
> Call-ID: 4f02699e2632410c359e1ee43a021dc7@18x.16.204.XXX:6060<br>
><br>
> CSeq: 102 OPTIONS<br>
><br>
><br>
> This is the peer.<br>
><br>
><br>
> * Name : desde-XopaXo-2376XXX<br>
> Secret : <Set><br>
> MD5Secret : <Not set><br>
> Remote Secret: <Not set><br>
> Context : from-trunk<br>
> Subscr.Cont. : <Not set><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 : *97<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 : No<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>
> DTMFmode : rfc2833<br>
> Timer T1 : 500<br>
> Timer B : 32000<br>
> ToHost : 201.217.31.10<br>
> Addr->IP : <a href="http://201.217.31.10:5060" target="_blank">201.217.31.10:5060</a><br>
> Defaddr->IP : (null)<br>
> Prim.Transp. : UDP<br>
> Allowed.Trsp : UDP<br>
> Def. Username: 595212376458<br>
> SIP Options : timer<br>
> Codecs : 0xe (gsm|ulaw|alaw)<br>
> Codec Order : (ulaw:20,alaw:20,gsm:20)<br>
> Auto-Framing : No<br>
> Status : OK (36 ms)<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>
> * Name : desde-XopaXo-2376XXX<br>
> Secret : <Set><br>
> MD5Secret : <Not set><br>
> Remote Secret: <Not set><br>
> Context : from-trunk<br>
> Subscr.Cont. : <Not set><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 : *97<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 : No<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>
> DTMFmode : rfc2833<br>
> Timer T1 : 500<br>
> Timer B : 32000<br>
> ToHost : 201.217.31.XX<br>
> Addr->IP : 201.217.31.XX:5060<br>
> Defaddr->IP : (null)<br>
> Prim.Transp. : UDP<br>
> Allowed.Trsp : UDP<br>
> Def. Username: 59X212376XXX<br>
> SIP Options : timer<br>
> Codecs : 0xe (gsm|ulaw|alaw)<br>
> Codec Order : (ulaw:20,alaw:20,gsm:20)<br>
> Auto-Framing : No<br>
> Status : OK (36 ms)<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>
><br>
><br>
><br>
<br>
<br>
<br>
</div></div><div class="HOEnZb"><div class="h5">--<br>
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</div></div></blockquote></div><br></div></div>