[asterisk-dev] [Code Review] Split user options from extension in chan_sip.

[jrose]

> On 2011-04-27 10:35:48, Russell Bryant wrote:
> > 1) There is a URI parsing unit test.  It would be good to add some test cases that exercise this code.
> > 
> > 2) The new CHANNEL() option needs to be added to the XML documentation.

The tests are proving somewhat difficult.  The parse_uri functions are left untouched by this change, so this patch really doesn't have any bearing on what happens in the tests you mentioned on IRC.  I'm not sure if I should be changing the parse_uri function to suit this change though since it is used outside of this context as well.

Working on adding the documentation now.


- jrose


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On 2011-04-20 15:10:16, jrose wrote:
> 
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> https://reviewboard.asterisk.org/r/1188/
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> 
> (Updated 2011-04-20 15:10:16)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> Adds a new option to SIP peers in order to truncate the semicolon delimited values in the URI so that devices which generate semicolon delimited values (like the Sonus mentioned in the bug report) can connect to an extension properly without forcing pattern matching and employing weird workarounds with the extension value every time it is used in the dialplan.
> 
> In order to add it to a peer, place the following in sip.conf:
> ignore_uri_sem=yes
> 
> By default it is off, naturally.
> 
> 
> This addresses bug 18344.
>     https://issues.asterisk.org/view.php?id=18344
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/channels/chan_sip.c 314156 
>   /branches/1.8/channels/sip/dialplan_functions.c 314156 
>   /branches/1.8/channels/sip/include/sip.h 314156 
> 
> Diff: https://reviewboard.asterisk.org/r/1188/diff
> 
> 
> Testing
> -------
> 
> Used sipp to make a phone call using the following xml:
> --------------------------------------
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
> 
> <scenario name="UAC OPTIONS">
>     <send retrans="500"> <![CDATA[
> 
>         INVITE sip:2005;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;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]
> 
>         ]]>
>     </send>
>     <recv response="200"/>
> </scenario>
> --------------------------------------
> 
> Tested those calls against dialplan as follows:
> --------------------------------------
> [sipp]
> exten => 2005,1,Answer()
> exten => 2005,2,Background(tt-weasels)
> exten => 2005,3,NoOp(callerid => ${CALLERID(all)})
> exten => 2005,n,NoOp(exten => ${EXTEN})
> exten => 2005,n,Wait(5)
> exten => 2005,n,HangUp()
> --------------------------------------
> 
> and used the following definitions of sipp in sip.conf:
> --------------------------------------
> [sipp]
> type=friend
> context=sipp
> host=dynamic
> port=6000
> user=sipp
> canreinvite=no
> diallow=all
> allow=ulaw
> ignore_uri_sem=on
> --------------------------------------
> 
> as well as ignore_uri_sem=off
> and with no ignore_uri_sem statement in the config file.
> 
> 
> Results:
> With the option on, the call gets completed as normal if the extension were just 2005.
> With the option off, the call gets interrupted with the following error message: 
> [Apr 18 14:40:58] NOTICE[16316]: chan_sip.c:21496 handle_request_invite: Call from 'sipp' to extension '2005' rejected because extension not found in context 'sipp'.
> 
> because the extension actually contains the entire URI and can't be matched in spite of it showing just 2005 above.
> 
> 
> Thanks,
> 
> jrose
> 
>

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