[asterisk-dev] Re: (IPv6) (punit)

Tue Sep 19 04:08:52 MST 2006

Hi

    i am new to asterisk but worked on IPv6 and want to work on IPv6 in asterisk.

   Can anyone help me how to understand the coding and start developing  
    new coding for IPv6

   Thanks in advance

  punit kandoi

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Today's Topics:

1. Re: using Hardware TDM switch (M Desai)
2. chan_skinny - call to turned off phone causes deadlock
(r43208) (Pavel Jezek)
3. Re: Bug 7966 RPID - works in SVN Trunk - What changed? (Steven)
4. Clarification on packetization feature (Dan Austin)
5. Re: Clarification on packetization feature (Matt O'Gorman)
6. SIP satck of asterisk (sudhir kumar)
7. Re: IPv6 (raman kumar)
8. Re: Re: IPv6 (Michiel van Baak)

----------------------------------------------------------------------

Message: 1
Date: Mon, 18 Sep 2006 12:02:30 -0700 (PDT)
From: M Desai 
Subject: Re: [asterisk-dev] using Hardware TDM switch
To: Asterisk Developers Mailing List 
Message-ID: <20060918190230.75244.qmail at web55103.mail.re4.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Thanks. I see how it works. I am hoping having a HW TDM switch makes things
a lot more deterministic and hopefully I can get rid of a lot of code.

One of the things I am playing around is for a secondary microcontroller to
control the POTS line cards (FXO and FXS) and connecting to the
main processor via a DP-RAM. The idea is that the signalling info
is filled in by the uC and the main processor just gets an interrupt to indicate
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
playing around and fortunately have acccess to a HW prototyping setup !

M Desai

Khelik Mikhail wrote: For this you should only correctly realize callback function which
will drive the hardware switch in your low-level driver and pass
pointer during the span registration, look into (*dacs) in the zt_span
structure.

2006/9/18, M Desai :
>
>
> I am trying out an old HW card which has one T1/E1 line, 8 POTS ports (4 FXO
> and 4 FXS) and a IDT TSI.I am trying to hack Zaptel to use the HW TSI so
> that only the signaling needs to be done in SW and the TDM channels for
> data will be handled in HW.
>
> Looks like I need only to change Zaptel to achieve this. Any advise on this
> from the gurus out there ? Is this feasible ?
>
> Also looks like the Octasic code attempts something along these lines but I
> presume the TDM switch is inside the Octasic 6100 can be used only to switch
> TDM channels inside the Echo canceller.
>
>
>
> M Desai
>
>
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Message: 2
Date: Mon, 18 Sep 2006 23:00:44 +0200
From: Pavel Jezek 

Subject: [asterisk-dev] chan_skinny - call to turned off phone causes
deadlock (r43208) 
To: asterisk-dev at lists.digium.com
Message-ID: <450F08FC.7030701 at i.cz>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hello,
I'm using 7920 registered to current asterisk svn version,
it working fine, but when I turn off phone, asterisk doesn't correctly 
handle unregister proces
when I call to this phone, I can hear normal ringback and asterisk 
deadlocks,
"stop now" does nothing, I must kill asterisk process.
I can see active call in "show channels" output, but without any 
details, like:

ipbx*CLI> show channels verbose
Channel Context Extension Prio State 
Application Data CallerID Duration 
Accountcode BridgedTo
0 active channels
3 active calls

normal status, phone ON and registered
ipbx*CLI> skinny show devices
Name DeviceId IP Type R NL
-------------------- ---------------- --------------- --------------- - --
PJ SEP000D288E664B 193.179.xxx.xxx 7920 Y 1

after turning phone OFF:

ipbx*CLI>
[Sep 18 22:50:22] WARNING[14186]: chan_skinny.c:4048 get_input: read() 
returned error: Connection reset by peer
[Sep 18 22:50:22] NOTICE[14186]: chan_skinny.c:4132 skinny_session: 
Skinny Session returned: Connection reset by peer

no IP is OK, but incorrect register status as "Y"?

ipbx*CLI>
ipbx*CLI> skinny show devices
Name DeviceId IP Type R NL
-------------------- ---------------- --------------- --------------- - --
PJ SEP000D288E664B 0.0.0.0 7920 Y 1

correct status as "N" after stop/start asterisk
ipbx*CLI> skinny show devices
Name DeviceId IP Type R NL
-------------------- ---------------- --------------- --------------- - --
PJ SEP000D288E664B No Device N 1

Asterisk SVN-trunk-r43208

------------------------------

Message: 3
Date: Mon, 18 Sep 2006 18:36:24 -0500
From: Steven 
Subject: Re: [asterisk-dev] Bug 7966 RPID - works in SVN Trunk - What
changed?
To: Asterisk Developers Mailing List 
Message-ID: <1158622584.20312.1.camel at bedroom.comcast.net>
Content-Type: text/plain

On Mon, 2006-09-18 at 10:47 -0700, Ed Greenberg wrote:
> Last night I filed bug 7966 against 1.2.12.1. Today, I find that the 
> problem is resolved as of SVN trunk r43075.
> 
> I would love to identify what changed between the two so I can patch my 
> 1.2.12.1 if possible and get past my XO usability test.
> 
> Can anybody assist me?

http://svn.digium.com/view/asterisk/trunk/apps/app_directory.c?rev=43075&r1=43074&r2=43075&view=diff

Wasn't that handy?
-- 
Steven 

------------------------------

Message: 4
Date: Mon, 18 Sep 2006 19:14:54 -0700
From: "Dan Austin" 
Subject: [asterisk-dev] Clarification on packetization feature
To: "Asterisk Developers Mailing List" 
Message-ID:

Content-Type: text/plain; charset="us-ascii"

I see that Mogorman has merged the packetization patch to trunk.
The svn commit comment is a little confusing, as there already is
a way to configure the local preference instead of reacting to the
remote endpoints request.

In chan_sip, chan_skinny and chan_ooh323 the ability to set per
codec packetization is in the allow= directive
(user/peer/friend/global):

allow=ulaw:30,g729:40,alaw

Will set ulaw to 30ms, G729 to 40ms and alaw to the default of 20ms

The default behaviour is to use the locally configured preferences
for all three channel drivers. Only in chan_sip is it optional to
also honor an endpoint request for a different packetization(ptime:).

So the work is a little more complete than the commit comment implies,
but perhaps under documented. I suppose a follow-up patch the adds
appropriate entries to the sample configs and something for the doc 
directory would be a good idea. I'll get something posted tomorrow
if no one beats me to it.

Dan

------------------------------

Message: 5
Date: Mon, 18 Sep 2006 23:11:51 -0500 (CDT)
From: "Matt O'Gorman" 
Subject: Re: [asterisk-dev] Clarification on packetization feature
To: Asterisk Developers Mailing List 
Message-ID:
<11138434.22281158639111210.JavaMail.root at jupiler.digium.com>
Content-Type: text/plain; charset=utf-8

Dan you are correct, disregard my commit message. everything is
perfect.

mog
----- Original Message -----
From: Dan Austin 
To: Asterisk Developers Mailing List 
Sent: Monday, September 18, 2006 9:14:54 PM GMT-0600
Subject: [asterisk-dev] Clarification on packetization feature

I see that Mogorman has merged the packetization patch to trunk.
The svn commit comment is a little confusing, as there already is
a way to configure the local preference instead of reacting to the
remote endpoints request.

In chan_sip, chan_skinny and chan_ooh323 the ability to set per
codec packetization is in the allow= directive
(user/peer/friend/global):

allow=ulaw:30,g729:40,alaw

Will set ulaw to 30ms, G729 to 40ms and alaw to the default of 20ms

The default behaviour is to use the locally configured preferences
for all three channel drivers. Only in chan_sip is it optional to
also honor an endpoint request for a different packetization(ptime:).

So the work is a little more complete than the commit comment implies,
but perhaps under documented. I suppose a follow-up patch the adds
appropriate entries to the sample configs and something for the doc 
directory would be a good idea. I'll get something posted tomorrow
if no one beats me to it.

Dan
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Message: 6
Date: Tue, 19 Sep 2006 05:21:49 +0100 (BST)
From: sudhir kumar 
Subject: [asterisk-dev] SIP satck of asterisk
To: asterisk-dev at lists.digium.com
Message-ID: <20060919042149.85233.qmail at web7612.mail.in.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1

Hi All,

I would like to know wether SIP stack of asterisk is
properitery or it own by Digium. 

Any lead is highly appericated. 

warmest regards
Sudhir

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------------------------------

Message: 7
Date: Tue, 19 Sep 2006 14:02:25 +0530
From: "raman kumar" 
Subject: [asterisk-dev] Re: IPv6
To: "Asterisk Developers Mailing List" 
Message-ID:
<67002eb30609190132k29d7844el88fe7e612c325c9a at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Yes I am also thinking for the same If u have started this astivity
then please tell me about the status so that I canalso contribute

On 17/09/06, Michiel van Baak wrote:
> Hi,
>
> I wonder if there's a developer working on IPv6 support?
> If not, is there any interest in someone working on ipv6
> support?
> I would be very happy if it's there, and since my C skills
> are being trained every day now I can help :)
>
> --
>
> Michiel van Baak
> michiel at vanbaak.eu
> http://michiel.vanbaak.eu
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
>
> "Why is it drug addicts and computer afficionados are both called users?"
>
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------------------------------

Message: 8
Date: Tue, 19 Sep 2006 11:43:58 +0200
From: Michiel van Baak 
Subject: Re: [asterisk-dev] Re: IPv6
To: asterisk-dev at lists.digium.com
Message-ID: <20060919094357.GB18901 at anima.vanbaak.info>
Content-Type: text/plain; charset=us-ascii

On 14:02, Tue 19 Sep 06, raman kumar wrote:
> Yes I am also thinking for the same If u have started this astivity
> then please tell me about the status so that I canalso contribute

I'll wait to see what Marc Blanchet comes up with.

-- 

Michiel van Baak
michiel at vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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