<div dir="ltr">Hi gurus!!!<br><br>I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn<br>Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions.<br>The every minute annoyng answer of the pstn is "403 Forbidden". <br>
Some people told that asterisk is not sending the username in the OPTION, required by the pstn.<br><br><br>Taking a look of the example of rfc3261.txt (pg 67), we found "carol", so it makingme see that i am missing some config.<br>
>><br> OPTIONS <a href="mailto:sip%3Acarol@chicago.com">sip:carol@chicago.com</a> SIP/2.0<br> Via: SIP/2.0/UDP <a href="http://pc33.atlanta.com">pc33.atlanta.com</a>;branch=z9hG4bKhjhs8ass877<br> Max-Forwards: 70<br>
To: <<a href="mailto:sip%3Acarol@chicago.com">sip:carol@chicago.com</a>><br><<<br><br><br>Is it wright?<br>How can i instruct FREEPBX to send the username in the option request?<br><br>Sorry for this silly question but a found no answer googling.<br>
<br><br><br>Thans in advance.<br>rv<br><br><br><br>This is the debug of the case<br><br><br>Reliably Transmitting (NAT) to 201.217.31.XX:5060:<br>OPTIONS sip:201.217.31.10 SIP/2.0<br>Via: SIP/2.0/UDP 18x.16.204.XXX:6060;branch=z9hG4bK1d8715df;rport<br>
Max-Forwards: 70<br>From: "Unknown" <sip:59X212376XXX@186.16.204.XXX:6060>;tag=as4491c6af<br>To: <sip:201.217.31.10><br>Contact: <sip:59X212376XXX@18x.16.204.XXX:6060><br>Call-ID: 4f02699e2632410c359e1ee43a021dc7@186.16.204.XXX:6060<br>
CSeq: 102 OPTIONS<br>User-Agent: FPBX-2.11.0(1.8.25.0)<br>Date: Wed, 25 Jun 2014 13:47:19 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Length: 0<br>
<br><br><--- SIP read from UDP:201.217.31.XX:5060 ---><br>SIP/2.0 403 Forbidden<br>Via: SIP/2.0/UDP 18x.16.204.XXX:6060;received=18x.16.204.XXX;branch=z9hG4bK1d8715df;rport=5060<br>From: "Unknown" <sip:59X212376XXX@18x.16.204.XXX:6060>;tag=as4491c6af<br>
To: <sip:201.217.31.XX>;tag=aprqngfrt-nm50ea10000c6<br>Call-ID: 4f02699e2632410c359e1ee43a021dc7@18x.16.204.XXX:6060<br><br>CSeq: 102 OPTIONS<br><br><br>This is the peer.<br><br><br> * Name : desde-XopaXo-2376XXX<br>
Secret : <Set><br> MD5Secret : <Not set><br> Remote Secret: <Not set><br> Context : from-trunk<br> Subscr.Cont. : <Not set><br> Language :<br> AMA flags : Unknown<br> Transfer mode: open<br>
CallingPres : Presentation Allowed, Not Screened<br> Callgroup :<br> Pickupgroup :<br> MOH Suggest :<br> Mailbox :<br> VM Extension : *97<br> LastMsgsSent : 32767/65535<br> Call limit : 0<br> Max forwards : 0<br>
Dynamic : No<br> Callerid : "" <><br> MaxCallBR : 384 kbps<br> Expire : -1<br> Insecure : port,invite<br> Force rport : Yes<br> ACL : No<br> DirectMedACL : No<br> T.38 support : No<br>
T.38 EC mode : Unknown<br> T.38 MaxDtgrm: -1<br> DirectMedia : No<br> PromiscRedir : No<br> User=Phone : No<br> Video Support: No<br> Text Support : No<br> Ign SDP ver : No<br> Trust RPID : No<br> Send RPID : No<br>
Subscriptions: Yes<br> Overlap dial : Yes<br> DTMFmode : rfc2833<br> Timer T1 : 500<br> Timer B : 32000<br> ToHost : 201.217.31.10<br> Addr->IP : <a href="http://201.217.31.10:5060">201.217.31.10:5060</a><br>
Defaddr->IP : (null)<br> Prim.Transp. : UDP<br> Allowed.Trsp : UDP<br> Def. Username: 595212376458<br> SIP Options : timer<br> Codecs : 0xe (gsm|ulaw|alaw)<br> Codec Order : (ulaw:20,alaw:20,gsm:20)<br>
Auto-Framing : No<br> Status : OK (36 ms)<br> Useragent :<br> Reg. Contact :<br> Qualify Freq : 60000 ms<br> Sess-Timers : Accept<br> Sess-Refresh : uas<br> Sess-Expires : 1800 secs<br> Min-Sess : 90 secs<br>
RTP Engine : asterisk<br> Parkinglot :<br> Use Reason : No<br> * Name : desde-XopaXo-2376XXX<br> Secret : <Set><br> MD5Secret : <Not set><br> Remote Secret: <Not set><br> Context : from-trunk<br>
Subscr.Cont. : <Not set><br> Language :<br> AMA flags : Unknown<br> Transfer mode: open<br> CallingPres : Presentation Allowed, Not Screened<br> Callgroup :<br> Pickupgroup :<br> MOH Suggest :<br>
Mailbox :<br> VM Extension : *97<br> LastMsgsSent : 32767/65535<br> Call limit : 0<br> Max forwards : 0<br> Dynamic : No<br> Callerid : "" <><br> MaxCallBR : 384 kbps<br> Expire : -1<br>
Insecure : port,invite<br> Force rport : Yes<br> ACL : No<br> DirectMedACL : No<br> T.38 support : No<br> T.38 EC mode : Unknown<br> T.38 MaxDtgrm: -1<br> DirectMedia : No<br> PromiscRedir : No<br>
User=Phone : No<br> Video Support: No<br> Text Support : No<br> Ign SDP ver : No<br> Trust RPID : No<br> Send RPID : No<br> Subscriptions: Yes<br> Overlap dial : Yes<br> DTMFmode : rfc2833<br> Timer T1 : 500<br>
Timer B : 32000<br> ToHost : 201.217.31.XX<br> Addr->IP : 201.217.31.XX:5060<br> Defaddr->IP : (null)<br> Prim.Transp. : UDP<br> Allowed.Trsp : UDP<br> Def. Username: 59X212376XXX<br> SIP Options : timer<br>
Codecs : 0xe (gsm|ulaw|alaw)<br> Codec Order : (ulaw:20,alaw:20,gsm:20)<br> Auto-Framing : No<br> Status : OK (36 ms)<br> Useragent :<br> Reg. Contact :<br> Qualify Freq : 60000 ms<br> Sess-Timers : Accept<br>
Sess-Refresh : uas<br> Sess-Expires : 1800 secs<br> Min-Sess : 90 secs<br> RTP Engine : asterisk<br> Parkinglot :<br> Use Reason : No<br><br></div>