[asterisk-users] OPTIONS Request without username <-> Forbidden
Brian LaVallee
b.lavallee at globaltank.jp
Thu Jul 3 04:18:05 CDT 2014
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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