[asterisk-dev] Re: asterisk-dev Digest, Vol 19, Issue 18

Sven Boeckelmann sboeckelmann at benelog.com
Tue Feb 7 04:25:35 MST 2006


ERLEDIGT:
XFree ist nun xorg



On Tue, 2006-02-07 at 01:30 -0700, asterisk-dev-request at lists.digium.com
wrote:
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> Today's Topics:
> 
>    1. Re: Distributed Asterisk (Stefan M?rkle)
>    2. Automerge changes... finally worked out (long) (Kevin P. Fleming)
>    3. Sound translation: asterisk-sounds archive too? (Philippe Lang)
>    4. Re: Sound translation: asterisk-sounds archive too?
>       (Michiel van Baak)
>    5. Zaptel 1.0 to latest version update (roswel ajf)
>    6. Re: Zaptel 1.0 to latest version update (Tilghman Lesher)
>    7. New issue tracker for handling licensing issues for Asterisk,
>       Zaptel and related projects (Asterisk Development Team)
>    8. Re: Sound translation: asterisk-sounds archive too?
>       (Tzafrir Cohen)
>    9. version.h in zaptel (Tzafrir Cohen)
>   10. SIP subscripion showing active for unavailable	channels
>       (Koopmann, Jan-Peter)
>   11. Re: app_meetme.c problem (Sergio Chersovani)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Mon, 06 Feb 2006 20:41:12 +0100
> From: Stefan M?rkle <stefan.maerkle at netpioneer.de>
> Subject: [asterisk-dev] Re: Distributed Asterisk
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <E1F6CEa-0007BB-UG at dolly.gnuher.de>
> Content-Type: text/plain; charset=ISO-8859-1
> 
> In article <43E77605.5070807 at caselaboratories.com> you wrote:
> > Are there any plans of making a distributed version of Asterisk?
> > A version where you can run line interface only on n computers and have 
> > the Call-Control/PABX on a sentralized one.
> > If so, any plans for redundancy on E1/T1's ?
> 
> Hmm. What exactly is it that IAX does? Or even TDMoE if you like?
> Nothing stops you from doing so, you don't need any
> multi-server-internal-communication overhead like corba or the like to
> achieve this goal. Good design helps.
> 
> Stefan
> 
> -- 
> Stefan Mrkle <stefan.maerkle at netpioneer.de>
> 
> 
> ------------------------------
> 
> Message: 2
> Date: Mon, 06 Feb 2006 15:00:46 -0600
> From: "Kevin P. Fleming" <kpfleming at digium.com>
> Subject: [asterisk-dev] Automerge changes... finally worked out (long)
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <43E7B8FE.2000104 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> I finally figured out the source of the problems with automerging of 
> developer branches from the trunk (this was not an issue with developer 
> branches based on branches/1.2).
> 
> The core issue that the 'svnmerge-integrated' property was being used 
> for two conflicting purposes: to track the branches/1.2 fixes that had 
> been forward-ported into the trunk and also to track the trunk changes 
> that had been merged into the developer branch. Obviously this cannot 
> work :-)
> 
> To solve the problem, the branches/1.2 forward-porting properties on 
> trunk are no longer using the standard svnmerge property names; they are 
> now called 'branch-1.2-merged' and 'branch-1.2-blocked', and there are 
> _no_ 'svnmerge-integrated' or 'svnmerge-blocked' properties on the 
> trunk. This means that forward-porting patches from branches/1.2 
> requires specifying the property names to svnmerge; see the 
> branching/merging page on asterisk.org for an example.
> 
> All new developers branches made from trunk should work with automerging 
> without any problem, but existing branches have an issue: the developer 
> branch contains a property 'svnmerge-integrated', but the trunk does 
> not, and running 'svnmerge merge' will try to update that property, 
> resulting in a conflict. To resolve this issue, for each branch that you 
> maintain, you will need to manually use svnmerge to bring it up-to-date 
> to at least revision 9163 (where the property was removed from the 
> trunk); from that point forward, automerge will be able to manage the 
> merges for you.
> 
> In the meantime, you will probably start to receive conflict notices 
> because your branches cannot be merged... but once we are past this 
> issue, automerging should work well and should no longer generate 
> 'false' conflicts like it was doing before.
> 
> 
> ------------------------------
> 
> Message: 3
> Date: Mon, 6 Feb 2006 22:25:13 +0100
> From: "Philippe Lang" <philippe.lang at attiksystem.ch>
> Subject: [asterisk-dev] Sound translation: asterisk-sounds archive
> 	too?
> To: <asterisk-dev at lists.digium.com>
> Message-ID:
> 	<6C0CF58A187DA5479245E0830AF84F4218CCAF at poweredge.attiksystem.ch>
> Content-Type: text/plain;	charset="utf-8"
> 
> Hi,
>  
> I intend to hire someone in order to translate Asterisk sounds in french. One question: is it sufficient to translate the *.gsm sounds of the asterisk-1.2.4.tar.gz archive, or is it also necessary to translate part of asterisk-sounds-1.2.1.tar.gz, so that all the Asterisk features work, like voicemailmain, etc...?
>  
> Thanks for the info
>  
> Philippe
> 
> ------------------------------
> 
> Message: 4
> Date: Mon, 6 Feb 2006 22:30:48 +0100
> From: Michiel van Baak <michiel at vanbaak.info>
> Subject: Re: [asterisk-dev] Sound translation: asterisk-sounds archive
> 	too?
> To: asterisk-dev at lists.digium.com
> Message-ID: <20060206213047.GG5288 at anima.vanbaak.info>
> Content-Type: text/plain; charset=us-ascii
> 
> On 22:25, Mon 06 Feb 06, Philippe Lang wrote:
> > Hi,
> >  
> > I intend to hire someone in order to translate Asterisk sounds in french. One question: is it sufficient to translate the *.gsm sounds of the asterisk-1.2.4.tar.gz archive, or is it also necessary to translate part of asterisk-sounds-1.2.1.tar.gz, so that all the Asterisk features work, like voicemailmain, etc...?
> 
> You only need to translate sounds from the asterisk-sounds
> pack if you are using them.
> Asterisk itself works great without them and will never use
> a file from that package if you don't tell it to :)
> 
> -- 
> Michiel van Baak
> http://michiel.vanbaak.info
> michiel at vanbaak.info
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
> 
> "Why is it drug addicts and computer afficionados are both called users?"
> 
> 
> 
> ------------------------------
> 
> Message: 5
> Date: Mon, 06 Feb 2006 22:13:07 +0000
> From: "roswel ajf" <roswel_ajf at hotmail.com>
> Subject: [asterisk-dev] Zaptel 1.0 to latest version update
> To: asterisk-dev at lists.digium.com
> Message-ID: <BAY101-F3125AB838CF955FB1E37F6ED0E0 at phx.gbl>
> Content-Type: text/plain; format=flowed
> 
> Hi,
> 
> We have the oldest (1.0) zaptel version runing with asterisk 1.0 version. I 
> am told to just upgrade teh zaptel only. 2 questions:
> 1. can i only upgrade zaptel to latest without upgrading the asterisk?
> 2. how can i upgrade such that, reverting back to the previous setup would 
> be simple?
> 
> thanks.
> 
> 
> 
> 
> ------------------------------
> 
> Message: 6
> Date: Mon, 6 Feb 2006 17:46:37 -0600
> From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
> Subject: Re: [asterisk-dev] Zaptel 1.0 to latest version update
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <200602061746.38653.tilghman at mail.jeffandtilghman.com>
> Content-Type: text/plain;  charset="iso-8859-6"
> 
> On Monday 06 February 2006 16:13, roswel ajf wrote:
> > We have the oldest (1.0) zaptel version runing with asterisk 1.0
> > version. I am told to just upgrade teh zaptel only. 2 questions:
> > 1. can i only upgrade zaptel to latest without upgrading the
> > asterisk? 2. how can i upgrade such that, reverting back to the
> > previous setup would be simple?
> 
> No, and no.  Thanks for playing and remember to take future questions
> like this to the -users list.
> 
> -- 
> Tilghman
> 
> 
> ------------------------------
> 
> Message: 7
> Date: Mon, 06 Feb 2006 17:59:28 -0600
> From: Asterisk Development Team <asteriskteam at digium.com>
> Subject: [asterisk-dev] New issue tracker for handling licensing
> 	issues for Asterisk, Zaptel and related projects
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <43E7E2E0.8080800 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> In an effort to ensure that every licensing issue brought to our
> attention is handled fully and openly, we have created a new issue
> tracker for this purpose. It is located at:
> 
> http://licensing.digium.com
> 
> The tracker is open to the public, and we encourage all interested
> parties to post their concerns and participate in the discussions
> involved in resolving them. Digium's will actively respond and pursue
> resolution of each and every issue posted, and other concerned parties
> may also participate.
> 
> In the future, if you have a question or concern related to the
> licensing of any of these projects (including packaging, trademark and
> other related issues), please open an issue in the tracker rather than
> posting to one of the mailing lists.
> 
> Thank you for supporting Asterisk and Zaptel!
> 
> 
> 
> ------------------------------
> 
> Message: 8
> Date: Tue, 7 Feb 2006 07:17:29 +0200
> From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
> Subject: Re: [asterisk-dev] Sound translation: asterisk-sounds archive
> 	too?
> To: asterisk-dev at lists.digium.com
> Message-ID: <20060207051729.GU16880 at xorcom.com>
> Content-Type: text/plain; charset=us-ascii
> 
> On Mon, Feb 06, 2006 at 10:25:13PM +0100, Philippe Lang wrote:
> > Hi,
> >  
> > I intend to hire someone in order to translate Asterisk sounds in 
> > french. 
> 
> BTW: anything wrong with the current french prompts?
> 
> -- 
> Tzafrir Cohen     icq#16849755  +972-50-7952406
> tzafrir.cohen at xorcom.com  http://www.xorcom.com
> 
> 
> ------------------------------
> 
> Message: 9
> Date: Tue, 7 Feb 2006 08:01:29 +0200
> From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
> Subject: [asterisk-dev] version.h in zaptel
> To: Asterisk Developers list <asterisk-dev at lists.digium.com>
> Message-ID: <20060207060129.GV16880 at xorcom.com>
> Content-Type: text/plain; charset=us-ascii
> 
> Hi
> 
> I noticed that zaptel tries to generate a version.h file. However when
> generating it from the tarball it seems to be empty, as no .version file
> is provided in the tarball.
> 
> Furthermore, I'm trying to understand what is it used for: a version of
> the kernel-user interface or for various zaptel modules to check on?
> Shouldn't all the other modules include it as well?
> 
> -- 
> Tzafrir Cohen     icq#16849755  +972-50-7952406
> tzafrir.cohen at xorcom.com  http://www.xorcom.com
> 
> 
> ------------------------------
> 
> Message: 10
> Date: Tue, 7 Feb 2006 08:17:14 +0100
> From: "Koopmann, Jan-Peter" <Jan-Peter.Koopmann at seceidos.de>
> Subject: [asterisk-dev] SIP subscripion showing active for unavailable
> 	channels
> To: <asterisk-dev at lists.digium.com>
> Message-ID:
> 	<AEF86EFA5497434190F6D57E2666EA7A41DB8E at ERWIN.intern.seceidos.de>
> Content-Type: text/plain;	charset="US-ASCII"
> 
> Hi,
> 
> not sure this is a bug or intended behaviour. Personally I think this is
> strange:
> 
> For some time I noticed that the subscription LEDs on my SNOM are on for
> extensions that are currently unavailable (example SIP phone not
> connected). Debugging the SIP Notification showed a "Subscription-State:
> active" SIP NOTIFY for those channels. This cannot be right, can it?
> 
> Similar thinks happen with Parking Slots using the patch from
> bugs.digium.org for parking slot hints.
> 
> Kind regards,
>   JP
> 
> 
> 
> 
> ------------------------------
> 
> Message: 11
> Date: Tue, 07 Feb 2006 09:30:03 +0100
> From: Sergio Chersovani <mlists at c-net.it>
> Subject: Re: [asterisk-dev] app_meetme.c problem
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <43E85A8B.8030400 at c-net.it>
> Content-Type: text/plain; charset=us-ascii; format=flowed
> 
> 
> >This is not only a problem with sccp.
> >One of our clients also has this with pure ZAP calls.
> >
> >Phonecalls come in on a sangoma 101 using latest wanpipe
> >drivers. asterisk 1.2.4
> >They also report the freezes when leaving meetme.
> >  
> >
> Please notify that to the bug tracker number 6422
> 
> Sergio
> 
> 
> ------------------------------
> 
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> End of asterisk-dev Digest, Vol 19, Issue 18
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