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 E.....@.9.9:=...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:ca...@chicago.com SIP/2.0 > Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877 > Max-Forwards: 70 > To: <sip:ca...@chicago.com> > << > > > 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:59x212376...@186.16.204.xxx:6060>;tag=as4491c6af > To: <sip:201.217.31.10> > Contact: <sip:59x212376...@18x.16.204.xxx:6060> > Call-ID: 4f02699e2632410c359e1ee43a021...@186.16.204.xxx: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:59x212376...@18x.16.204.xxx:6060>;tag=as4491c6af > To: <sip:201.217.31.XX>;tag=aprqngfrt-nm50ea10000c6 > Call-ID: 4f02699e2632410c359e1ee43a021...@18x.16.204.xxx: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