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