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[visser.rafael at gmail...]

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@chicago.com ([email]sip%3Acarol@chicago.com[/email]) SIP/2.0
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
      Max-Forwards: 70
      To: <sip:carol@chicago.com ([email]sip%3Acarol@chicago.com[/email])>
<<


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@186.16.204.XXX:6060>;tag=as4491c6af
To: <sip:201.217.31.10>
Contact: <sip:59X212376XXX@18x.16.204.XXX:6060>
Call-ID: 4f02699e2632410c359e1ee43a021dc7@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@18x.16.204.XXX:6060>;tag=as4491c6af
To: <sip:201.217.31.XX>;tag=aprqngfrt-nm50ea10000c6
Call-ID: 4f02699e2632410c359e1ee43a021dc7@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

[rnewton at digium.com]

On Wed, Jun 25, 2014 at 9:30 AM, Rafael Visser <visser.rafael@gmail.com> wrote:

[b.lavallee at globalta...]

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:

[visser.rafael at gmail...]

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@globaltank.jp (b.lavallee@globaltank.jp)>: