<div>Hi</div> <div> </div> <div> i am new to asterisk but worked on IPv6 and want to work on IPv6 in asterisk.</div> <div> </div> <div> Can anyone help me how to understand the coding and start developing </div> <div> new coding for IPv6</div> <div> </div> <div> Thanks in advance</div> <div> </div> <div>punit kandoi<BR><BR><B><I>asterisk-dev-request@lists.digium.com</I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Send asterisk-dev mailing list submissions to<BR>asterisk-dev@lists.digium.com<BR><BR>To subscribe or unsubscribe via the World Wide Web, visit<BR>http://lists.digium.com/mailman/listinfo/asterisk-dev<BR>or, via email, send a message with subject or body 'help' to<BR>asterisk-dev-request@lists.digium.com<BR><BR>You can reach the person managing the list at<BR>asterisk-dev-owner@lists.digium.com<BR><BR>When replying,
please edit your Subject line so it is more specific<BR>than "Re: Contents of asterisk-dev digest..."<BR><BR><BR>Today's Topics:<BR><BR>1. Re: using Hardware TDM switch (M Desai)<BR>2. chan_skinny - call to turned off phone causes deadlock<BR>(r43208) (Pavel Jezek)<BR>3. Re: Bug 7966 RPID - works in SVN Trunk - What changed? (Steven)<BR>4. Clarification on packetization feature (Dan Austin)<BR>5. Re: Clarification on packetization feature (Matt O'Gorman)<BR>6. SIP satck of asterisk (sudhir kumar)<BR>7. Re: IPv6 (raman kumar)<BR>8. Re: Re: IPv6 (Michiel van Baak)<BR><BR><BR>----------------------------------------------------------------------<BR><BR>Message: 1<BR>Date: Mon, 18 Sep 2006 12:02:30 -0700 (PDT)<BR>From: M Desai <MDESAI_1970@YAHOO.COM><BR>Subject: Re: [asterisk-dev] using Hardware TDM switch<BR>To: Asterisk Developers Mailing List <ASTERISK-DEV@LISTS.DIGIUM.COM><BR>Message-ID: <20060918190230.75244.qmail@web55103.mail.re4.yahoo.com><BR>Content-Type:
text/plain; charset="iso-8859-1"<BR><BR>Thanks. I see how it works. I am hoping having a HW TDM switch makes things<BR>a lot more deterministic and hopefully I can get rid of a lot of code.<BR><BR><BR>One of the things I am playing around is for a secondary microcontroller to<BR>control the POTS line cards (FXO and FXS) and connecting to the<BR>main processor via a DP-RAM. The idea is that the signalling info<BR>is filled in by the uC and the main processor just gets an interrupt to indicate<BR>signalling data is available in the DP-RAM. It is an overkill (but I am curious to see if my setup can hadle FXS port in the 400-800 range) but as I said I am just<BR>playing around and fortunately have acccess to a HW prototyping setup !<BR><BR>M Desai<BR><BR><BR>Khelik Mikhail <MIXEL.NET@GMAIL.COM>wrote: For this you should only correctly realize callback function which<BR>will drive the hardware switch in your low-level driver and pass<BR>pointer during the span registration,
look into (*dacs) in the zt_span<BR>structure.<BR><BR>2006/9/18, M Desai :<BR>><BR>><BR>> I am trying out an old HW card which has one T1/E1 line, 8 POTS ports (4 FXO<BR>> and 4 FXS) and a IDT TSI.I am trying to hack Zaptel to use the HW TSI so<BR>> that only the signaling needs to be done in SW and the TDM channels for<BR>> data will be handled in HW.<BR>><BR>> Looks like I need only to change Zaptel to achieve this. Any advise on this<BR>> from the gurus out there ? Is this feasible ?<BR>><BR>> Also looks like the Octasic code attempts something along these lines but I<BR>> presume the TDM switch is inside the Octasic 6100 can be used only to switch<BR>> TDM channels inside the Echo canceller.<BR>><BR>><BR>><BR>> M Desai<BR>><BR>><BR>> ________________________________<BR>> How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call rates.<BR>><BR>><BR>>
_______________________________________________<BR>> --Bandwidth and Colocation provided by Easynews.com --<BR>><BR>> asterisk-dev mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-dev<BR>><BR>><BR>><BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>asterisk-dev mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-dev<BR><BR><BR><BR>---------------------------------<BR>How low will we go? Check out Yahoo! Messenger’s low PC-to-Phone call rates.<BR>-------------- next part --------------<BR>An HTML attachment was scrubbed...<BR>URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20060918/93bf8032/attachment-0001.htm<BR><BR>------------------------------<BR><BR>Message: 2<BR>Date: Mon, 18 Sep 2006 23:00:44 +0200<BR>From: Pavel Jezek
<PAVEL.JEZEK@I.CZ><BR>Subject: [asterisk-dev] chan_skinny - call to turned off phone causes<BR>deadlock (r43208) <BR>To: asterisk-dev@lists.digium.com<BR>Message-ID: <450F08FC.7030701@i.cz><BR>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<BR><BR>Hello,<BR>I'm using 7920 registered to current asterisk svn version,<BR>it working fine, but when I turn off phone, asterisk doesn't correctly <BR>handle unregister proces<BR>when I call to this phone, I can hear normal ringback and asterisk <BR>deadlocks,<BR>"stop now" does nothing, I must kill asterisk process.<BR>I can see active call in "show channels" output, but without any <BR>details, like:<BR><BR>ipbx*CLI> show channels verbose<BR>Channel Context Extension Prio State <BR>Application Data CallerID Duration <BR>Accountcode BridgedTo<BR>0 active channels<BR>3 active calls<BR><BR><BR><BR><BR>normal status, phone ON and registered<BR>ipbx*CLI> skinny show devices<BR>Name DeviceId IP Type R
NL<BR>-------------------- ---------------- --------------- --------------- - --<BR>PJ SEP000D288E664B 193.179.xxx.xxx 7920 Y 1<BR><BR><BR><BR>after turning phone OFF:<BR><BR>ipbx*CLI><BR>[Sep 18 22:50:22] WARNING[14186]: chan_skinny.c:4048 get_input: read() <BR>returned error: Connection reset by peer<BR>[Sep 18 22:50:22] NOTICE[14186]: chan_skinny.c:4132 skinny_session: <BR>Skinny Session returned: Connection reset by peer<BR><BR>no IP is OK, but incorrect register status as "Y"?<BR><BR>ipbx*CLI><BR>ipbx*CLI> skinny show devices<BR>Name DeviceId IP Type R NL<BR>-------------------- ---------------- --------------- --------------- - --<BR>PJ SEP000D288E664B 0.0.0.0 7920 Y 1<BR><BR><BR><BR>correct status as "N" after stop/start asterisk<BR>ipbx*CLI> skinny show devices<BR>Name DeviceId IP Type R NL<BR>-------------------- ---------------- --------------- --------------- - --<BR>PJ SEP000D288E664B No Device N 1<BR><BR><BR>Asterisk
SVN-trunk-r43208<BR><BR><BR>------------------------------<BR><BR>Message: 3<BR>Date: Mon, 18 Sep 2006 18:36:24 -0500<BR>From: Steven <CRITCH@BASESYS.COM><BR>Subject: Re: [asterisk-dev] Bug 7966 RPID - works in SVN Trunk - What<BR>changed?<BR>To: Asterisk Developers Mailing List <ASTERISK-DEV@LISTS.DIGIUM.COM><BR>Message-ID: <1158622584.20312.1.camel@bedroom.comcast.net><BR>Content-Type: text/plain<BR><BR>On Mon, 2006-09-18 at 10:47 -0700, Ed Greenberg wrote:<BR>> Last night I filed bug 7966 against 1.2.12.1. Today, I find that the <BR>> problem is resolved as of SVN trunk r43075.<BR>> <BR>> I would love to identify what changed between the two so I can patch my <BR>> 1.2.12.1 if possible and get past my XO usability test.<BR>> <BR>> Can anybody assist me?<BR><BR>http://svn.digium.com/view/asterisk/trunk/apps/app_directory.c?rev=43075&r1=43074&r2=43075&view=diff<BR><BR>Wasn't that handy?<BR>-- <BR>Steven
<CRITCH@BASESYS.COM><BR><BR><BR><BR>------------------------------<BR><BR>Message: 4<BR>Date: Mon, 18 Sep 2006 19:14:54 -0700<BR>From: "Dan Austin" <DAN_AUSTIN@PHOENIX.COM><BR>Subject: [asterisk-dev] Clarification on packetization feature<BR>To: "Asterisk Developers Mailing List" <ASTERISK-DEV@LISTS.DIGIUM.COM><BR>Message-ID:<BR><B0CF4196F21DC0448367514774331AB7019FDD17@SCL-EXCH2K3.PHOENIX.COM><BR>Content-Type: text/plain; charset="us-ascii"<BR><BR>I see that Mogorman has merged the packetization patch to trunk.<BR>The svn commit comment is a little confusing, as there already is<BR>a way to configure the local preference instead of reacting to the<BR>remote endpoints request.<BR><BR>In chan_sip, chan_skinny and chan_ooh323 the ability to set per<BR>codec packetization is in the allow= directive<BR>(user/peer/friend/global):<BR><BR>allow=ulaw:30,g729:40,alaw<BR><BR>Will set ulaw to 30ms, G729 to 40ms and alaw to the default of 20ms<BR><BR>The default behaviour is to use
the locally configured preferences<BR>for all three channel drivers. Only in chan_sip is it optional to<BR>also honor an endpoint request for a different packetization(ptime:).<BR><BR>So the work is a little more complete than the commit comment implies,<BR>but perhaps under documented. I suppose a follow-up patch the adds<BR>appropriate entries to the sample configs and something for the doc <BR>directory would be a good idea. I'll get something posted tomorrow<BR>if no one beats me to it.<BR><BR>Dan<BR><BR><BR>------------------------------<BR><BR>Message: 5<BR>Date: Mon, 18 Sep 2006 23:11:51 -0500 (CDT)<BR>From: "Matt O'Gorman" <MOGORMAN@DIGIUM.COM><BR>Subject: Re: [asterisk-dev] Clarification on packetization feature<BR>To: Asterisk Developers Mailing List <ASTERISK-DEV@LISTS.DIGIUM.COM><BR>Message-ID:<BR><11138434.22281158639111210.JavaMail.root@jupiler.digium.com><BR>Content-Type: text/plain; charset=utf-8<BR><BR>Dan you are correct, disregard my commit
message. everything is<BR>perfect.<BR><BR>mog<BR>----- Original Message -----<BR>From: Dan Austin <DAN_AUSTIN@PHOENIX.COM><BR>To: Asterisk Developers Mailing List <ASTERISK-DEV@LISTS.DIGIUM.COM><BR>Sent: Monday, September 18, 2006 9:14:54 PM GMT-0600<BR>Subject: [asterisk-dev] Clarification on packetization feature<BR><BR>I see that Mogorman has merged the packetization patch to trunk.<BR>The svn commit comment is a little confusing, as there already is<BR>a way to configure the local preference instead of reacting to the<BR>remote endpoints request.<BR><BR>In chan_sip, chan_skinny and chan_ooh323 the ability to set per<BR>codec packetization is in the allow= directive<BR>(user/peer/friend/global):<BR><BR>allow=ulaw:30,g729:40,alaw<BR><BR>Will set ulaw to 30ms, G729 to 40ms and alaw to the default of 20ms<BR><BR>The default behaviour is to use the locally configured preferences<BR>for all three channel drivers. Only in chan_sip is it optional to<BR>also honor an endpoint
request for a different packetization(ptime:).<BR><BR>So the work is a little more complete than the commit comment implies,<BR>but perhaps under documented. I suppose a follow-up patch the adds<BR>appropriate entries to the sample configs and something for the doc <BR>directory would be a good idea. I'll get something posted tomorrow<BR>if no one beats me to it.<BR><BR>Dan<BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>asterisk-dev mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-dev<BR><BR><BR><BR>------------------------------<BR><BR>Message: 6<BR>Date: Tue, 19 Sep 2006 05:21:49 +0100 (BST)<BR>From: sudhir kumar <SUDHIR_KUMAR_T@YAHOO.CO.IN><BR>Subject: [asterisk-dev] SIP satck of asterisk<BR>To: asterisk-dev@lists.digium.com<BR>Message-ID: <20060919042149.85233.qmail@web7612.mail.in.yahoo.com><BR>Content-Type: text/plain;
charset=iso-8859-1<BR><BR><BR>Hi All,<BR><BR>I would like to know wether SIP stack of asterisk is<BR>properitery or it own by Digium. <BR><BR><BR>Any lead is highly appericated. <BR><BR>warmest regards<BR>Sudhir<BR><BR><BR><BR><BR>__________________________________________________________<BR>Yahoo! India Answers: Share what you know. Learn something new<BR>http://in.answers.yahoo.com/<BR><BR><BR>------------------------------<BR><BR>Message: 7<BR>Date: Tue, 19 Sep 2006 14:02:25 +0530<BR>From: "raman kumar" <RAMANK24@GMAIL.COM><BR>Subject: [asterisk-dev] Re: IPv6<BR>To: "Asterisk Developers Mailing List" <ASTERISK-DEV@LISTS.DIGIUM.COM><BR>Message-ID:<BR><67002eb30609190132k29d7844el88fe7e612c325c9a@mail.gmail.com><BR>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<BR><BR>Yes I am also thinking for the same If u have started this astivity<BR>then please tell me about the status so that I canalso contribute<BR><BR><BR>On 17/09/06, Michiel van Baak
<MICHIEL@VANBAAK.INFO>wrote:<BR>> Hi,<BR>><BR>> I wonder if there's a developer working on IPv6 support?<BR>> If not, is there any interest in someone working on ipv6<BR>> support?<BR>> I would be very happy if it's there, and since my C skills<BR>> are being trained every day now I can help :)<BR>><BR>> --<BR>><BR>> Michiel van Baak<BR>> michiel@vanbaak.eu<BR>> http://michiel.vanbaak.eu<BR>> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD<BR>><BR>> "Why is it drug addicts and computer afficionados are both called users?"<BR>><BR>> _______________________________________________<BR>> --Bandwidth and Colocation provided by Easynews.com --<BR>><BR>> asterisk-dev mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-dev<BR>><BR><BR><BR>------------------------------<BR><BR>Message: 8<BR>Date: Tue, 19 Sep 2006 11:43:58
+0200<BR>From: Michiel van Baak <MICHIEL@VANBAAK.INFO><BR>Subject: Re: [asterisk-dev] Re: IPv6<BR>To: asterisk-dev@lists.digium.com<BR>Message-ID: <20060919094357.GB18901@anima.vanbaak.info><BR>Content-Type: text/plain; charset=us-ascii<BR><BR>On 14:02, Tue 19 Sep 06, raman kumar wrote:<BR>> Yes I am also thinking for the same If u have started this astivity<BR>> then please tell me about the status so that I canalso contribute<BR><BR>I'll wait to see what Marc Blanchet comes up with.<BR><BR>-- <BR><BR>Michiel van Baak<BR>michiel@vanbaak.eu<BR>http://michiel.vanbaak.eu<BR>GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD<BR><BR>"Why is it drug addicts and computer afficionados are both called users?"<BR><BR><BR><BR>------------------------------<BR><BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>asterisk-dev mailing list<BR>To UNSUBSCRIBE or update options
visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-dev<BR><BR><BR>End of asterisk-dev Digest, Vol 26, Issue 77<BR>********************************************<BR></BLOCKQUOTE><BR><p> 
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