Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] How to check if a SIP phone is forwardedwit


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
steve.langstaff at cit...
Guest





PostPosted: Tue Jan 08, 2008 8:11 am    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <kpfleming at digium.com>:


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)




-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080108/111ea983/attachment.htm
Back to top
oza-4h07 at myamail.com
Guest





PostPosted: Tue Jan 08, 2008 9:31 am    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

2008/1/8, Steve Langstaff <steve.langstaff at citel.com>:
Quote:

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.

------------------------------
Quote:
*From:* asterisk-users-bounces at lists.digium.com [mailto:
asterisk-users-bounces at lists.digium.com] *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 <kpfleming at digium.com>:
Quote:

Olivier wrote:

Quote:
Is there way for an Asterisk server to check if a sip phone is
forwarded
Quote:
without bothering phone's user ?

No.

Quote:
I was thinking of some Alert-Info option that would let the phone
reply
Quote:
with a 302 Moved Temporarily or 182 Queued message and not let the
phone
Quote:
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

--
Quote:
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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080108/0f28b46d/attachment.htm
Back to top
rj2807 at gmail.com
Guest





PostPosted: Tue Jan 08, 2008 8:48 pm    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <kpfleming at digium.com>:

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)
Back to top
oej at edvina.net
Guest





PostPosted: Wed Jan 09, 2008 1:50 am    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

9 jan 2008 kl. 02.48 skrev Raj Jain:

Quote:
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
Back to top
benny+usenet at amorse...
Guest





PostPosted: Wed Jan 09, 2008 4:53 am    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

Olivier <oza-4h07 at myamail.com> writes:

Quote:
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
Back to top
oza-4h07 at myamail.com
Guest





PostPosted: Wed Jan 09, 2008 12:53 pm    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

2008/1/9, Benny Amorsen <benny+usenet at amorsen.dk>:
Quote:

Olivier <oza-4h07 at myamail.com> writes:

Quote:
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 ? Wink

/Benny
Quote:



_______________________________________________
-- 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080109/d06b9d5e/attachment.htm
Back to top
benny+usenet at amorse...
Guest





PostPosted: Thu Jan 10, 2008 2:45 am    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

Olivier <oza-4h07 at myamail.com> writes:

Quote:
To get a polite "go to hell !" in return ? Wink

I think the vendors will be nicer than that. Asterisk has a good bit
of the VoIP PBX market.
/Benny
Back to top
oza-4h07 at myamail.com
Guest





PostPosted: Thu Jan 10, 2008 5:01 am    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

2008/1/10, Benny Amorsen <benny+usenet at amorsen.dk>:
Quote:

Olivier <oza-4h07 at myamail.com> writes:

Quote:
To get a polite "go to hell !" in return ? Wink

I think the vendors will be nicer than that.
You're right.

Asterisk has a good bit
Quote:
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
Quote:


_______________________________________________
-- 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/b9a2db22/attachment.htm
Back to top
oza-4h07 at myamail.com
Guest





PostPosted: Mon Jan 14, 2008 8:32 am    Post subject: [asterisk-users] How to check if a SIP phone is forwardedwit Reply with quote

2008/1/9, Johansson Olle E <oej at edvina.net>:
Quote:


9 jan 2008 kl. 02.48 skrev Raj Jain:

Quote:
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
?3.5.2) using STUN technique is recommended.
Do you foresee phone manufacturers to support this ?


/O
Quote:

_______________________________________________
-- 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080114/dd437612/attachment.htm
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services