[asterisk-dev] [Code Review] Yet Another Whack at the SIP user options issue.

[jrose]

> On 2011-05-19 14:12:22, jrose wrote:
> > lmadsen: jrose: what are the chances you could make the option name something like:  legacy_useroption_parsing={yes,no} or something like that?

That's fine by me.


- jrose


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On 2011-05-19 14:02:21, jrose wrote:
> 
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> 
> (Updated 2011-05-19 14:02:21)
> 
> 
> Review request for Asterisk Developers, Russell Bryant, David Vossel, and Leif Madsen.
> 
> 
> Summary
> -------
> 
> A quick recap:
> As of Asterisk 1.8, semicolons in user fields are accepted as just part of the user field in compliance with RFC 3261.  This makes devices that employ those options unable to register and also unable to match intended extensions without dialplan workarounds.
> 
> This approach involves a global sip option (if it were done per channel, we couldn't match on registers unfortunately) to strip the semicolons in the same general way as it was done in 1.6.2.  For that reason I chose to call it legacyuseroptionparsing
> 
> 
> This addresses bug 18344.
>     https://issues.asterisk.org/view.php?id=18344
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/channels/chan_sip.c 319526 
>   /branches/1.8/channels/sip/include/sip.h 319526 
>   /branches/1.8/configs/sip.conf.sample 319526 
> 
> Diff: https://reviewboard.asterisk.org/r/1223/diff
> 
> 
> Testing
> -------
> 
> Made sure matches while the option were on would happen with the following using sipp:
> 
> 
>         REGISTER sip:localhost SIP/2.0
>         Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> 		To: "Jonathan" <sip:evilhost;garbage@[local_ip]:[local_port]>
>         From: "Jonathan" <sip:evilhost;garbage@[local_ip]:[local_port]>;tag=[call_number]
> 
>         Call-ID: [call_id]
>         CSeq: 1 REGISTER
>         Contact: sip:evilhost@[local_ip]:[local_port];expires=3600
>         ALLOW: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> 		User-Agent: Sipp
>         Content-Length: [len]
> 
> And
> 
> 
>         INVITE sip:2005;5002;phone-context=+1;npdi=yes@[remote_ip]:[remote_port] SIP/2.0
>         Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>         From: "Lrrrr Schmrrr" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
>         To: Asterisk <sip:2005;5002;phone-context=+1;npdi=yes@[remote_ip]:[remote_port]>
>         Call-ID: [call_id]
>         CSeq: 1 OPTIONS
>         Contact: sip:sipp@[local_ip]:[local_port]
>         Max-Forwards: 70
>         Subject: Asterisk Testsuite
>         Content-Length: [len]
> 
> 
> Also that they also acted the same as the way they acted before with the option off.
> 
> 
> Thanks,
> 
> jrose
> 
>

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