Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
2008/1/9, Johansson Olle E [EMAIL PROTECTED]: 9 jan 2008 kl. 02.48 skrev Raj Jain: This issue of phone vendors not supporting OPTIONS according to RFC 3261 often comes up on this list. Like Kevin Fleming said, an OPTIONS request is supposed to be responded in the same way as an INVITE. Almost all SIP phone vendors have construed OPTIONS as some kind of a keep-alive request, which is wrong. Which we do too, by the way. In worst case, maybe Asterisk has set this industry standard. OPTIONS is far to heavy in processing on the server side to be used for keep-alives. I'm starting to see devices that use it for checking capabilities - the proper way. To do this properly, we will have to authenticate the OPTIONs request and match it with the proper peer/ user to get the proper codec settings, ACLs and such. Since all versions of Asterisk use OPTIONs for NAT-keepalives, I'm a bit hesitant to fix this. It's a catch 22. I want to do it properly, but then the amount of processing for each OPTIONs request that we receive is going to be a bit too much. Maybe one could ask vendors to add a header to the OPTIONs packet saying this is just a keep-alive. Give me a 200 OK without any parsing and be happy, because I don't care about the reply. Linksys has a setting and use NOTIFY for Keep-alives, which also is a poor solution, but at least something we can just give an error response to without a lot of processing. There was a proposal for PING, but it never got anywhere. Here (http://www3.tools.ietf.org/html/draft-ietf-sip-outbound-11#page-11 ยง3.5.2) using STUN technique is recommended. Do you foresee phone manufacturers to support this ? /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
2008/1/10, Benny Amorsen [EMAIL PROTECTED]: Olivier [EMAIL PROTECTED] writes: To get a polite go to hell ! in return ? ;-) I think the vendors will be nicer than that. You're right. Asterisk has a good bit of the VoIP PBX market. Asking all of them for guidance (how do you plan to change your phone firmware for keeping NAT alive ?) within the same letter, would help to get an answer from those who haven't decided yet which way to follow or don't rate this question with a high priority (I think the majority of phone manufacturers are in this case : maintaining a dual OPTIONS behaviour for today's and future's interoperability isn't something they would be happy to support). Asking all of them would also give the impression the prospective market is larger than Asterisk's market share. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
Olivier [EMAIL PROTECTED] writes: As using OPTIONS requests main benefit is to non-phone specific, what shall we do when most vendors do not comply with RFC ? Write polite letters to the vendors? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
2008/1/9, Benny Amorsen [EMAIL PROTECTED]: Olivier [EMAIL PROTECTED] writes: As using OPTIONS requests main benefit is to non-phone specific, what shall we do when most vendors do not comply with RFC ? Write polite letters to the vendors? To get a polite go to hell ! in return ? ;-) /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
Olivier [EMAIL PROTECTED] writes: To get a polite go to hell ! in return ? ;-) I think the vendors will be nicer than that. Asterisk has a good bit of the VoIP PBX market. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
That's going to be pretty phone-specific. How about asking your phone supplier to fix their phone so that it responds to OPTIONS correctly? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: 08 January 2008 12:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ? 2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]: Olivier wrote: Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? No. I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. According to the SIP RFC, a SIP endpoint is supposed to respond to an OPTIONS message the same way that it would respond to an INVITE message with the identical destination, but I've never seen a phone respond to an OPTIONS message with anything but '200 OK', even when a redirect (forward) is in place. So, the alternative option is to play with html and use phone embedded html server to get this redirection data. Cheers -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
2008/1/8, Steve Langstaff [EMAIL PROTECTED]: That's going to be pretty phone-specific. How about asking your phone supplier to fix their phone so that it responds to OPTIONS correctly? Yes, you're right but RFC3261 doesn't specify such 302 replies. So I'm very pessimistic about my phone supplier answer. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Olivier *Sent:* 08 January 2008 12:50 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ? 2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]: Olivier wrote: Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? No. I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. According to the SIP RFC, a SIP endpoint is supposed to respond to an OPTIONS message the same way that it would respond to an INVITE message with the identical destination, but I've never seen a phone respond to an OPTIONS message with anything but '200 OK', even when a redirect (forward) is in place. So, the alternative option is to play with html and use phone embedded html server to get this redirection data. Cheers -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
This issue of phone vendors not supporting OPTIONS according to RFC 3261 often comes up on this list. Like Kevin Fleming said, an OPTIONS request is supposed to be responded in the same way as an INVITE. Almost all SIP phone vendors have construed OPTIONS as some kind of a keep-alive request, which is wrong. Can we ask the phone vendors to play by the book? -- Raj From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Tuesday, January 08, 2008 7:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ? 2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]: Olivier wrote: Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? No. I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. According to the SIP RFC, a SIP endpoint is supposed to respond to an OPTIONS message the same way that it would respond to an INVITE message with the identical destination, but I've never seen a phone respond to an OPTIONS message with anything but '200 OK', even when a redirect (forward) is in place. So, the alternative option is to play with html and use phone embedded html server to get this redirection data. Cheers -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
As using OPTIONS requests main benefit is to non-phone specific, what shall we do when most vendors do not comply with RFC ? 2008/1/9, Raj Jain [EMAIL PROTECTED]: This issue of phone vendors not supporting OPTIONS according to RFC 3261 often comes up on this list. Like Kevin Fleming said, an OPTIONS request is supposed to be responded in the same way as an INVITE. Almost all SIP phone vendors have construed OPTIONS as some kind of a keep-alive request, which is wrong. Can we ask the phone vendors to play by the book? -- Raj From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Tuesday, January 08, 2008 7:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ? 2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]: Olivier wrote: Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? No. I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. According to the SIP RFC, a SIP endpoint is supposed to respond to an OPTIONS message the same way that it would respond to an INVITE message with the identical destination, but I've never seen a phone respond to an OPTIONS message with anything but '200 OK', even when a redirect (forward) is in place. So, the alternative option is to play with html and use phone embedded html server to get this redirection data. Cheers -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?
9 jan 2008 kl. 02.48 skrev Raj Jain: This issue of phone vendors not supporting OPTIONS according to RFC 3261 often comes up on this list. Like Kevin Fleming said, an OPTIONS request is supposed to be responded in the same way as an INVITE. Almost all SIP phone vendors have construed OPTIONS as some kind of a keep-alive request, which is wrong. Which we do too, by the way. In worst case, maybe Asterisk has set this industry standard. OPTIONS is far to heavy in processing on the server side to be used for keep-alives. I'm starting to see devices that use it for checking capabilities - the proper way. To do this properly, we will have to authenticate the OPTIONs request and match it with the proper peer/ user to get the proper codec settings, ACLs and such. Since all versions of Asterisk use OPTIONs for NAT-keepalives, I'm a bit hesitant to fix this. It's a catch 22. I want to do it properly, but then the amount of processing for each OPTIONs request that we receive is going to be a bit too much. Maybe one could ask vendors to add a header to the OPTIONs packet saying this is just a keep-alive. Give me a 200 OK without any parsing and be happy, because I don't care about the reply. Linksys has a setting and use NOTIFY for Keep-alives, which also is a poor solution, but at least something we can just give an error response to without a lot of processing. There was a proposal for PING, but it never got anywhere. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
