[asterisk-dev] [Code Review] Add interoperability friendly option to SIP peers in order to cull semicolon delimited values from the URI

[jrose]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1188/
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(Updated 2011-04-19 12:08:57.702240)


Review request for Asterisk Developers.


Changes
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No longer optional.
User options are now stored in an ast_string_field within the pvt (uri_user_options)
channel function can now use "uri_user_options" as the argument to retrieve the options string from the pvt string field above.

Also the splitting is now performed in an axillary function and I've removed some stylistic changes since I was no longer 'in the neighborhood' so to speak.


Summary
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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 (updated)
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  /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
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Used sipp to make a phone call using the following xml:
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<?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:
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[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:
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[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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