Re: [asterisk-users] OPTIONS Request without username - Forbidden
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:212555@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:212555@PROXY_IP:5060;tag=gK094dc1e4 To: sip:212555@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-nm50ea1c6 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 : 6 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 :
Re: [asterisk-users] OPTIONS Request without username - Forbidden
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.laval...@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.@.9.9:=...j.p.n$BINVITE sip:212555@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:212555@PROXY_IP:5060;tag=gK094dc1e4 To: sip:212555@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-nm50ea1c6 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 : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk
Re: [asterisk-users] OPTIONS Request without username - Forbidden
On Wed, Jun 25, 2014 at 9:30 AM, Rafael Visser visser.raf...@gmail.com 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. Is it wright? How can i instruct FREEPBX to send the username in the option request? It may be worth asking on the FreePBX forums at http://community.freepbx.org/ as the Asterisk users who use FreePBX are generally monitoring that community. Many people here won't be able to answer your question *within the context* of FreePBX configuration. Your question is also not clear. You should ask the provider specifically which header and where in what URI they want to see the username in. If this wasn't FreePBX I'd tell you to just try setting the callerid and fromuser options for the corresponding SIP peer. I don't want to pretend to know FreePBX, so I still recommend you go ask on their forum to get better assistance. Good luck! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] OPTIONS Request without username - Forbidden
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-nm50ea1c6 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 : 6 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 : 6 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:
