[asterisk-users] OPTIONS Request without username <-> Forbidden
Rafael Visser
visser.rafael at gmail.com
Thu Jul 3 09:49:00 CDT 2014
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 <b.lavallee at globaltank.jp>:
> 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..... at .9.9:=...j.p".....n$BINVITE sip:2125551111 at 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 at PROXY_IP:5060>;tag=gK094dc1e4
> To: <sip:2125551111 at CARRIER_IP:5060>;tag=as2953dd14
> Call-ID: 1980326667_35899190 at 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:carol at chicago.com SIP/2.0
> > Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
> > Max-Forwards: 70
> > To: <sip:carol at 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:59X212376XXX at 186.16.204.XXX:6060>;tag=as4491c6af
> > To: <sip:201.217.31.10>
> > Contact: <sip:59X212376XXX at 18x.16.204.XXX:6060>
> > Call-ID: 4f02699e2632410c359e1ee43a021dc7 at 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:59X212376XXX at 18x.16.204.XXX:6060>;tag=as4491c6af
> > To: <sip:201.217.31.XX>;tag=aprqngfrt-nm50ea10000c6
> > Call-ID: 4f02699e2632410c359e1ee43a021dc7 at 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
> >
> >
> >
>
>
>
> --
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