Re: [asterisk-dev] Clearing pick-up groups on Zap/ channels
On Nov 8, 2006, at 9:44 AM, Kevin P. Fleming wrote: Tzafrir Cohen wrote: Note that I don't aim that high. I just aim at keeping this option open (let alone making 'reload chan_zap.so' work as defined). As the US Air Force recruitment ads say: AIM HIGH! Seriously... lots of people will be very happy if someone out there can find the time to re-do chan_zap's config system to make it compatible with the rest of Asterisk, compatible with Realtime (for those who want to use it) and easier to manage from automated maintenance tools like GUIs. It's on our internal list of projects, but it's there with 30+ other things that are probably going to happen first... Yeah, to be honest, I've been saving such major changes to config files and such for whenever I can start working on a new version of chan_zap that would replace it (or supersede it) for some configurations. There is a lot of reorganizational work that should be done, and personally I think we need to break the CCS signalling protocols (ISDN/SS7) out into another channel driver. There are things more important than the config file format that I would like to see change in chan_zap. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?
Luigi Rizzo wrote:
> i.e. tell it what extra libs we need.
> I modified the macro (which is "our code", in acinclude.m4)
> to support multiple instances and stop at the
> first matching one e.g.
>
> AST_EXT_LIB_CHECK([TONEZONE], [tonezone], [tone_zone_find],
> [zaptel/tonezone.h], [${tonezone_extra}], [140])
> # other case, old tonezone (0.80)
> AST_EXT_LIB_CHECK([TONEZONE], [tonezone], [tone_zone_find],
> [zaptel/zaptel.h], [${tonezone_extra}], [80])
Ahh, OK. That makes sense then.
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[asterisk-dev] configuration of chan_zap
Hi What do we want to see in the configuration of chan_zap? If we want to configure it just like any other channel type. So for starters, let's consider the standard configuration format: one section per object. What objects do we have here? * Analog channels * Groups of analog channels * Digital channels * Digital spans * Span groups (I don't know those well enough). If we consider analopg channels or groups of them as objects, the model works well. As for digital channels and digital spans: I don't know this well enough. As it seems that this will require breaking the format, let's just call the new format, should it be required, zap.conf. What I'm asking myself is how to remove the current limitation on the reload function of chan_zap: that there won't be substatial configuration changes. That is: no channels are added/removed or having their signalling changed. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] SIP Multiple endpoints with same id
On Wed, Nov 08, 2006 at 09:53:55AM -0600, Kevin P. Fleming wrote: > Alexei Volkov wrote: > > Is it possible (in theory) to make asterisk server multiple sip endponts > > configured with same sip credentials. > > Of course it's possible (in theory). Asterisk is software, software can > be programmed to do anything people want it to do. > > > > > If asterisk can support multiple sip enpoints with same credentials i > > have simplify routint to them with simple Dial(SIP/0001) instead of > > Dial(SIP/0001&SIP/0002&SIP/0003) command and save dialplan numbers (one > > instead of three in my case). > > > > In spite of the fact that it is possible, we have no plans nor desire to > make this change. Asterisk treats SIP devices as _devices_, which means > it needs to be able to handle them individually. Turning devices into > _extensions_ is the responsibility of the dialplan, where it is very > very easy to do what you want to do (as your example shows). The not-so-easy part (to me at least) is how to make this dynamic - e.g. buy a new _device_, configure with appropriate credentials, and have it dynamically (i.e. as it registers/unregisters) linked to a certain extension together with the other devices in the group without having to modify the dialplan. If there is a well known trick to achieve this, it would be surely worthwhile documenting it in sip.conf or some other prominent place. It would cut a lot of questions. cheers luigi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Asterisk 1.4 and Queues RealTime
Hi all, I would like to use the Agent Login feature with real-time queues … it is not possible with asterisk 1.2, as described here : http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue “The mantis bug describing the implementation of realtime queue is bug 4037. This bug includes some discussion on how to extend dynamic queues to also work with the member login feature.” So, did you know if this is possible “natively” in asterisk 1.4 (or will be) ? Thanks Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Clearing pick-up groups on Zap/ channels
On Wed, Nov 08, 2006 at 09:44:54AM -0600, Kevin P. Fleming wrote: > Tzafrir Cohen wrote: > > Note that I don't aim that high. I just aim at keeping this option open > > (let alone making 'reload chan_zap.so' work as defined). > > As the US Air Force recruitment ads say: AIM HIGH! But first I'd like to fix bugs. The current implementation is buggy, and I have suggested a simple way to fix it. It has some other nice side effect for future modifications, but my short-term aim is simpler. The long-term goal and short-term goal are not exactly conflicting here. > > Seriously... lots of people will be very happy if someone out there can > find the time to re-do chan_zap's config system to make it compatible > with the rest of Asterisk, compatible with Realtime (for those who want > to use it) and easier to manage from automated maintenance tools like > GUIs. It's on our internal list of projects, but it's there with 30+ > other things that are probably going to happen first... Naturally... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] aclocal.m4 ???
Kevin P. Fleming wrote: I think we should do that, especially because it will reduce the size of the generated aclocal.m4 (and potentially the configure script itself, depending on how smart m4 is). Alright, I'll work on doing that right now. However, I don't think the size of aclocal.m4 really matters since that has been removed from the tree. I'll do it for the sake of making things easier on the people that have to regenerate the configure script. :) -- Russell Bryant Software Engineer Digium, Inc. begin:vcard fn:Russell Bryant n:Bryant;Russell org:Digium, Inc. adr:;;150 West Park Loop;Huntsville;AL;35806;USA email;internet:[EMAIL PROTECTED] title:Software Engineer tel;work:+1-256-428-6000 x-mozilla-html:FALSE url:http://www.digium.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?
On Wed, Nov 08, 2006 at 09:46:39AM -0600, Kevin P. Fleming wrote:
> Luigi Rizzo wrote:
> > well, the thing is, on FreeBSD at least libnsl does not exist so
> > putting it in the loader flags breaks the build.
> > The same may happen for other platforms.
> > i would suggest to remove these extra libs (except for gnutls, but probably
> > it
> > is already there after detecting gnutls) and let autoconf handle the
> > dependency.
>
> autoconf doesn't 'handle dependencies' on its own; it does only what we
> tell it to do.
what i meant was the following (examples from our configure.ac):
if AST_EXT_LIB_SETUP([IKSEMEL], [Iksemel Jabber Library], [iksemel])
says yes, it means that iksemel itself links without extra libs,
otherwise we would have used something like this
AST_EXT_LIB_CHECK([OSPTK], [osptk], [OSPPCryptoDecrypt], [osp/osp.h], [-lcrypto
-lssl])
i.e. tell it what extra libs we need.
I modified the macro (which is "our code", in acinclude.m4)
to support multiple instances and stop at the
first matching one e.g.
AST_EXT_LIB_CHECK([TONEZONE], [tonezone], [tone_zone_find],
[zaptel/tonezone.h], [${tonezone_extra}], [140])
# other case, old tonezone (0.80)
AST_EXT_LIB_CHECK([TONEZONE], [tonezone], [tone_zone_find], [zaptel/zaptel.h],
[${tonezone_extra}], [80])
cheers
luigi
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Re: [asterisk-dev] SIP Multiple endpoints with same id
Alexei Volkov wrote:
> Is it possible (in theory) to make asterisk server multiple sip endponts
> configured with same sip credentials.
Of course it's possible (in theory). Asterisk is software, software can
be programmed to do anything people want it to do.
> If asterisk can support multiple sip enpoints with same credentials i
> have simplify routint to them with simple Dial(SIP/0001) instead of
> Dial(SIP/0001&SIP/0002&SIP/0003) command and save dialplan numbers (one
> instead of three in my case).
>
In spite of the fact that it is possible, we have no plans nor desire to
make this change. Asterisk treats SIP devices as _devices_, which means
it needs to be able to handle them individually. Turning devices into
_extensions_ is the responsibility of the dialplan, where it is very
very easy to do what you want to do (as your example shows).
Part of your problem is that you are making an assumption that your SIP
device identifiers have to be the same format (and come from the same
number space) as your extensions ("dialplan numbers"). This is a faulty
assumption; they are completely unrelated and generally using the same
numbers in both places only causes confusion. Your SIP devices can be
called anything you want (many people use device MAC addresses or
portions of them for hardphones, for example), and the extensions can be
numeric as you have shown.
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Re: [asterisk-dev] aclocal.m4 ???
Russell Bryant wrote: > That macro is provided when you install libtool. We can work around > that dependency by copying the macro into our file that contains our > custom macros, acinclude.m4 if we would like. I think we should do that, especially because it will reduce the size of the generated aclocal.m4 (and potentially the configure script itself, depending on how smart m4 is). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Why -lnsl in IKSEMEL_LIB ?
Luigi Rizzo wrote: > well, the thing is, on FreeBSD at least libnsl does not exist so > putting it in the loader flags breaks the build. > The same may happen for other platforms. > i would suggest to remove these extra libs (except for gnutls, but probably it > is already there after detecting gnutls) and let autoconf handle the > dependency. autoconf doesn't 'handle dependencies' on its own; it does only what we tell it to do. I'll ping Matt today to figure out whether we really need this at all on Linux, although yesterday he told me he's very, very close to having the new version of iksemel ready that uses OpenSSL and this would be a moot point anyway. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Clearing pick-up groups on Zap/ channels
Tzafrir Cohen wrote: > Note that I don't aim that high. I just aim at keeping this option open > (let alone making 'reload chan_zap.so' work as defined). As the US Air Force recruitment ads say: AIM HIGH! Seriously... lots of people will be very happy if someone out there can find the time to re-do chan_zap's config system to make it compatible with the rest of Asterisk, compatible with Realtime (for those who want to use it) and easier to manage from automated maintenance tools like GUIs. It's on our internal list of projects, but it's there with 30+ other things that are probably going to happen first... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk 1.4b3 crashing consistantly - How do I provide helpful info?
On Tue, 2006-11-07 at 22:33 -0500, Greg Boehnlein wrote: > Two things you need to make sure you enable when you compile asterisk: > > DEBUG_THREADS > DONT_OPTIMIZE > > These can be toggled under the "Compiler Flags" options when you do a > "make menuselect". Make sure that you have them turned on (I.E. an > Asterisk next to them). > > Then, follow the guidelines for obtaining a backtrace with GDB. These can > be found in the file "asterisk/doc/backtrace.txt". > > I would suggest that you issue the following, to make your life easy; > > script backtrace.txt <- This will start a typescript session, which > captures the Input / Output from the session into > a file called "backtrace.txt" > > Once you get into gdb, then issue: > "set pagination off" to turn of paging. > > Then, get your backtrace, exit out of GDB and come find someone on > #asterisk-dev on irc.freenode.org to take a look at your backtrace. > Thank you. I hope you don't mind but I have updated the wiki with those tips for others. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Re: [zaptel-commits] file: trunk r1570 - /trunk/wct4xxp/Kbuild
Kevin P. Fleming wrote: Actually, we need to do this all the way back to branch-1.2 as well... and ensure that we aren't building the firmware into the module unless absolutely necessary. Now that there are two firmware blobs, it will make the module even larger :-( That is already taken care of in all of them. It was just building the firmware utility and header files needlessly, not really "hurting" anything but it was wasting time. I wouldn't exactly call it a critical bugfix for things, thus why I didn't do it back further. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Re: [zaptel-commits] file: trunk r1570 - /trunk/wct4xxp/Kbuild
[EMAIL PROTECTED] wrote: > Author: file > Date: Tue Nov 7 23:08:31 2006 > New Revision: 1570 > > URL: http://svn.digium.com/view/zaptel?rev=1570&view=rev > Log: > Don't build the firmware headers unless needed. This shaves ~3.5 seconds off > build time. Actually, we need to do this all the way back to branch-1.2 as well... and ensure that we aren't building the firmware into the module unless absolutely necessary. Now that there are two firmware blobs, it will make the module even larger :-( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] SIP Multiple endpoints with same id
No it is not possible. Alexei Volkov wrote: Is it possible (in theory) to make asterisk server multiple sip endponts configured with same sip credentials. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] SIP Multiple endpoints with same id
Hi all! Is it possible (in theory) to make asterisk server multiple sip endponts configured with same sip credentials. For example if i have in sip.conf [0001] canreinvite=yes type=friend secret=secret username=USER callerid="USER" <0001> host=dynamic Is it possible to have two or more hardware or software sip phones configured with this entry? I tested that asterisk allow to register more then one endpoint with same credentials and allow outgoinc calls from them, but will route incoming call to last registered one. Why not to keep information about all such sip endpoints and route incoming calls to all devices with same sip credentials. This situation is helpfull when i have more then one sip phones (for example softphone on pc SIP/0001, hardware desktop sip phone SIP/0002, and mobile wifi sip phone SIP/0003) and nowtime i need to configure them with different sip numbers (and configuration entry in sip.conf) and have a special dialplan in case i want to route any call to me on all this devices. If asterisk can support multiple sip enpoints with same credentials i have simplify routint to them with simple Dial(SIP/0001) instead of Dial(SIP/0001&SIP/0002&SIP/0003) command and save dialplan numbers (one instead of three in my case). Thanks for any comments. WBR Alexei Volkov. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
