[asterisk-dev] asterisk-dev Digest, Vol 81, Issue 105
Riduan Shah
syncbyte at me.com
Mon May 2 21:33:29 CDT 2011
hello,
i need help on my Freepbx,
On 28-Apr-2011, at 1:00 AM, asterisk-dev-request at lists.digium.com wrote:
> Send asterisk-dev mailing list submissions to
> asterisk-dev at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> or, via email, send a message with subject or body 'help' to
> asterisk-dev-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-dev-owner at lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-dev digest..."
>
>
> Today's Topics:
>
> 1. Re: [Code Review] Split user options from extension in
> chan_sip. (Russell Bryant)
> 2. Re: [Code Review] Changes to h323 to allow use with h323plus
>> 1.20. (irroot)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 27 Apr 2011 15:35:48 -0000
> From: "Russell Bryant" <reviewboard at asterisk.org>
> Subject: Re: [asterisk-dev] [Code Review] Split user options from
> extension in chan_sip.
> To: , "Russell Bryant" <reviewboard at asterisk.org>, "Asterisk
> Developers" <asterisk-dev at lists.digium.com>
> Message-ID: <20110427153548.30328.36058 at hotblack.digium.com>
> Content-Type: text/plain; charset="utf-8"
>
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1188/#review3444
> -----------------------------------------------------------
>
>
> 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.
>
> - Russell
>
>
> On 2011-04-20 15:10:16, jrose wrote:
>>
>> -----------------------------------------------------------
>> This is an automatically generated e-mail. To reply, visit:
>> https://reviewboard.asterisk.org/r/1188/
>> -----------------------------------------------------------
>>
>> (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
>>
>>
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20110427/b00ab111/attachment-0001.htm>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 27 Apr 2011 15:53:06 -0000
> From: "irroot" <reviewboard at asterisk.org>
> Subject: Re: [asterisk-dev] [Code Review] Changes to h323 to allow use
> with h323plus > 1.20.
> To: "irroot" <reviewboard at asterisk.org>, , "Asterisk Developers"
> <asterisk-dev at lists.digium.com>, "Paul Belanger"
> <pabelanger at digium.com>
> Message-ID: <20110427155306.31854.89853 at hotblack.digium.com>
> Content-Type: text/plain; charset="utf-8"
>
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1117/
> -----------------------------------------------------------
>
> (Updated 2011-04-27 10:53:06.784628)
>
>
> Review request for Asterisk Developers.
>
>
> Changes
> -------
>
> Applied additional formating.
>
>
> Summary
> -------
>
> Changes to h323 to allow use with h323plus > 1.20.
>
>
> This addresses bugs 16619 and 18670.
> https://issues.asterisk.org/view.php?id=16619
> https://issues.asterisk.org/view.php?id=18670
>
>
> Diffs (updated)
> -----
>
> /trunk/channels/chan_h323.c 315724
> /trunk/channels/h323/ast_h323.h 315724
> /trunk/channels/h323/ast_h323.cxx 315724
> /trunk/channels/h323/chan_h323.h 315724
> /trunk/configs/h323.conf.sample 315724
> /trunk/configure.ac 315724
>
> Diff: https://reviewboard.asterisk.org/r/1117/diff
>
>
> Testing
> -------
>
>
> Thanks,
>
> irroot
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20110427/7745fab9/attachment-0001.htm>
>
> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2010 - October 26-28 Washington, DC
> Put in your talk proposal: http://www.bit.ly/speak-astricon2010
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> End of asterisk-dev Digest, Vol 81, Issue 105
> *********************************************
More information about the asterisk-dev
mailing list