Re: [asterisk-dev] [asterisk-commits] rizzo: trunk r89452 - in /trunk: acinclude.m4 configure configure.ac
In article <[EMAIL PROTECTED]>, Simon Perreault <[EMAIL PROTECTED]> wrote: > On Wednesday 21 November 2007 09:57:54 Patrick wrote: > > Similarly I really don't understand why Asterisk 1.4 requires autoconf > > 2.60 when that is not supported by the distros mentioned above. One > > would think this is taken into account. Funny thing is that after > > ripping out some parts, bootstrap.sh works fine on CentOS 5 with > > autoconf 2.59. So why hardcoding 2.60? But I digress. > > It's because of AC_USE_SYSTEM_EXTENSIONS. It's only available starting with > 2.60. Is that an essential feature to use, or could whatever it is used for be achieved in some more compatible way? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [asterisk-commits] kpfleming: trunk r89481 - /trunk/channels/chan_usbradio.c
On Wed, Nov 21, 2007 at 03:45:56PM -, SVN commits to the Asterisk project
wrote:
> Author: kpfleming
> Date: Wed Nov 21 09:45:56 2007
> New Revision: 89481
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=89481
> Log:
> get this to actually compile...
interesting... another client for my config parsing macros :)
cheers
luigi
> +++ trunk/channels/chan_usbradio.c Wed Nov 21 09:45:56 2007
> @@ -224,7 +224,7 @@
> * Likely we will come up with a better way of doing config file parsing.
> */
> #define M_START(var, val) \
> -char *__s = var; char *__val = val;
> +const char *__s = var; const char *__val = val;
> #define M_END(x) x;
> #define M_F(tag, f) if (!strcasecmp((__s), tag)) { f; } else
> #define M_BOOL(tag, dst) M_F(tag, (dst) = ast_true(__val) )
> @@ -505,7 +505,7 @@
>
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Multiple realm in SIP channel
I'm checking possibility of serving multiple realms with one Asterisk.
When the call comes to Asterisk it responds with authorization request
using specified realm (global_realm in code).
OpenSER for this case has possibility to auto generate realm based on
"From" header:
if (!proxy_authorize("", "subscriber)) {
proxy_challenge("", "1"); # Realm will be autogenerated
};
proxy_authorize(realm, table)
* realm - Realm is a opaque string that the user agent
should present to the user so he can decide what username
and password to use. Usually this is domain of the host
the server is running on.
If an empty string "" is used then the server will
generate it from the request. From header field domain
will be used as realm.
The string may contain pseudo variables.
Is it a good idea to add the same possibility to Asterisk?
E.g. possibility to generate Realm from "From" may be enabled in sip.conf with
additional list of acceptable realms.
Victor
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [asterisk-commits] rizzo: tru nk r89452 - in /trunk: acinclude.m4 configu re configure.ac
On Wednesday 21 November 2007 09:57:54 Patrick wrote: > Similarly I really don't understand why Asterisk 1.4 requires autoconf > 2.60 when that is not supported by the distros mentioned above. One > would think this is taken into account. Funny thing is that after > ripping out some parts, bootstrap.sh works fine on CentOS 5 with > autoconf 2.59. So why hardcoding 2.60? But I digress. It's because of AC_USE_SYSTEM_EXTENSIONS. It's only available starting with 2.60. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [asterisk-commits] rizzo: trunk r89452 - in /trunk: acinclude.m4 configure configure.ac
On Tue, 2007-11-20 at 13:28 -0500, Simon Perreault wrote: > On Tuesday 20 November 2007 13:20:49 Luigi Rizzo wrote: > > it is actually a big burden. there are 145 different version of each of the > > tools, some of which are incompatible with each other, > > and not all build on all platforms. > > This statement sounds awfully like FUD. > > What's wrong with autoconf >= 2.60? That's the official guideline, hardcoded > into configure.ac. Pretty much any distribution nowadays comes with it. Nothing except that Red Hat RHEL 4.x and 5, CentOS 4.x and 5 all have 2.59 or earlier so enforcing 2.60 will cause problems for the users of these distros that want/need to run bootstrap.sh. Imho "pretty much" is a whole lot smaller with these 2 distros not in the autoconf 2.60 camp. Similarly I really don't understand why Asterisk 1.4 requires autoconf 2.60 when that is not supported by the distros mentioned above. One would think this is taken into account. Funny thing is that after ripping out some parts, bootstrap.sh works fine on CentOS 5 with autoconf 2.59. So why hardcoding 2.60? But I digress. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Deprecating sip call-limit
Olle E Johansson wrote: > 20 nov 2007 kl. 22.58 skrev BJ Weschke: > > >> Johansson Olle E wrote: >> >>> Friends, >>> >>> Blitzrage and I had a discussion about busylevel and call-limit in >>> chan_sip on the IRC I wanted to expand to the rest of you deveopers >>> out there... >>> >>> >>> My proposal in this discussion was: >>> >>> - deprecate call-limit in chan_sip. No other channel driver has a >>> built-in call limit >>>and we now favour groupcount and dialplan control instead of >>> embedding this >>>into channel drivers. >>>The call-limit is history that has survived too long. >>> >>> - implement a new option to enable call counters for the subscribe/ >>> notify event system in chan_sip (channel specific) >>> >>> - implement busy-level in more channel drivers >>> >>> - implement a DEVICE() dial plan function that is cross-channel, like >>> CHANNEL(), >>> so we can check busylevel in chan_iax2 and other channels that can >>> handle multiple >>>channels per device. Busylevel can now be checked in the SIPPEER() >>> function only. >>> >>> >>> I know this is a lot of stuff at the same time, but it kind of >>> belongs >>> together. >>> >>> /O >>> >>> >> Olle / Leif - >> >> What are we going to do about things like app_queue that resolve on >> call-limit and limitonpeers being set correctly in order to make a >> proper response back to app_queue that the device is or is not busy >> to receive a queue call? >> > > App_queue doesn't do anything with the actual limit, but it needs the > call counter that is the basis behind the call limit. That counter > will stay, but get a new name. The actual enforcement of any call > limits in chan_sip will go. > > Ah! Ok. In that case, then yes, +1. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Need
then go to voip-info home page... > sir i need a starting tutorial > > Mayank Mathur <[EMAIL PROTECTED]> wrote: > so what prob r u facing > >> it could be any application . >> >> Mayank Mathur wrote: >> hi >> what appl r u developing?? is it IVR based appl ? >> >> >> >>> Hi to all >>> >>> I have just started to development small application in asterisk ..and >>> while reading "app_skel.c" many things confused me ...could any on >>> suggest me from where to start or is there any usefull tutorial >>> ...thanks >>> >>> >>> - >>> Get easy, one-click access to your favorites. Make Yahoo! your >>> homepage.___ >>> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >>> >>> asterisk-dev mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-dev >> >> >> -- >> >> Regards, >> Mayank Mathur >> >> >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-dev >> >> >> >> - >> Get easy, one-click access to your favorites. Make Yahoo! your >> homepage.___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-dev > > > -- > > Regards, > Mayank Mathur > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > - > Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try > it now.___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Regards, Mayank Mathur ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Need
sir i need a starting tutorial Mayank Mathur <[EMAIL PROTECTED]> wrote: so what prob r u facing > it could be any application . > > Mayank Mathur wrote: > hi > what appl r u developing?? is it IVR based appl ? > > > >> Hi to all >> >> I have just started to development small application in asterisk ..and >> while reading "app_skel.c" many things confused me ...could any on >> suggest me from where to start or is there any usefull tutorial >> ...thanks >> >> >> - >> Get easy, one-click access to your favorites. Make Yahoo! your >> homepage.___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-dev > > > -- > > Regards, > Mayank Mathur > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > - > Get easy, one-click access to your favorites. Make Yahoo! your > homepage.___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Regards, Mayank Mathur ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev - Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Need
sir i need a starting tutorial Mayank Mathur <[EMAIL PROTECTED]> wrote: so what prob r u facing > it could be any application . > > Mayank Mathur wrote: > hi > what appl r u developing?? is it IVR based appl ? > > > >> Hi to all >> >> I have just started to development small application in asterisk ..and >> while reading "app_skel.c" many things confused me ...could any on >> suggest me from where to start or is there any usefull tutorial >> ...thanks >> >> >> - >> Get easy, one-click access to your favorites. Make Yahoo! your >> homepage.___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-dev > > > -- > > Regards, > Mayank Mathur > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > - > Get easy, one-click access to your favorites. Make Yahoo! your > homepage.___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Regards, Mayank Mathur ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev - Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Need
so what prob r u facing > it could be any application . > > Mayank Mathur <[EMAIL PROTECTED]> wrote: > hi > what appl r u developing?? is it IVR based appl ? > > > >> Hi to all >> >> I have just started to development small application in asterisk ..and >> while reading "app_skel.c" many things confused me ...could any on >> suggest me from where to start or is there any usefull tutorial >> ...thanks >> >> >> - >> Get easy, one-click access to your favorites. Make Yahoo! your >> homepage.___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-dev > > > -- > > Regards, > Mayank Mathur > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > - > Get easy, one-click access to your favorites. Make Yahoo! your > homepage.___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Regards, Mayank Mathur ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Need
it could be any application . Mayank Mathur <[EMAIL PROTECTED]> wrote: hi what appl r u developing?? is it IVR based appl ? > Hi to all > > I have just started to development small application in asterisk ..and > while reading "app_skel.c" many things confused me ...could any on > suggest me from where to start or is there any usefull tutorial ...thanks > > > - > Get easy, one-click access to your favorites. Make Yahoo! your > homepage.___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Regards, Mayank Mathur ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev - Get easy, one-click access to your favorites. Make Yahoo! your homepage.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] dialplan redesign - don't forget characters (New topic)
21 nov 2007 kl. 10.01 skrev Luigi Rizzo: > On Wed, Nov 21, 2007 at 09:28:55AM +0100, Olle E Johansson wrote: >>> >>> Really, this is not an area where you can afford playing and putting >>> in >>> small patches to see how they fix one or the other problem. >>> The correctness of extension matching is something that people >>> really must rely on, because in the end it is directly involved >>> with security, billing and so on. >>> >> One thing that has to be considered as well if we redesign is >> alphanumeric > > There is nothing preventing alphanumeric extensions at least in the > old matching algorithm - the only annoyance is the need to 'escape' > some characters ( _ N Z X / come to mind ) as [N] [Z] etc. to > override their special meaning on asterisk patterns. Well, you can't properly match [EMAIL PROTECTED] today. Or give ranges like [A-Ö]xp[0-3] > But then, this is trivial to overcome by providing an alternative > syntax for extensions (and supporting both in parallel is easy) > and besides all regexp implementations have their own special > characters. That was our proposal. Let's find a URL so you can check it. http://edvina.net/asterisk/alphanumericextensions.pdf May 2005 :-) TTT - Things Take Time... /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Need
hi what appl r u developing?? is it IVR based appl ? > Hi to all > > I have just started to development small application in asterisk ..and > while reading "app_skel.c" many things confused me ...could any on > suggest me from where to start or is there any usefull tutorial ...thanks > > > - > Get easy, one-click access to your favorites. Make Yahoo! your > homepage.___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Regards, Mayank Mathur ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Need
Hi to all I have just started to development small application in asterisk ..and while reading "app_skel.c" many things confused me ...could any on suggest me from where to start or is there any usefull tutorial ...thanks - Get easy, one-click access to your favorites. Make Yahoo! your homepage.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] dialplan redesign - don't forget characters (New topic)
On Wed, Nov 21, 2007 at 09:28:55AM +0100, Olle E Johansson wrote: > > > > Really, this is not an area where you can afford playing and putting > > in > > small patches to see how they fix one or the other problem. > > The correctness of extension matching is something that people > > really must rely on, because in the end it is directly involved > > with security, billing and so on. > > > One thing that has to be considered as well if we redesign is > alphanumeric There is nothing preventing alphanumeric extensions at least in the old matching algorithm - the only annoyance is the need to 'escape' some characters ( _ N Z X / come to mind ) as [N] [Z] etc. to override their special meaning on asterisk patterns. But then, this is trivial to overcome by providing an alternative syntax for extensions (and supporting both in parallel is easy) and besides all regexp implementations have their own special characters. cheers luigi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [asterisk-commits] rizzo: trunk r89452 -?in?/trunk: acinclude.m4 configure configure.ac
On Tue, Nov 20, 2007 at 11:56:16PM -0600, Jeffrey Ollie wrote: > On 11/20/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > > > Anybody tried to work with git-svn with the Asterisk repository? > > Yeah, it works pretty well... I use it to maintain the various patches > that I apply to my Fedora packages. You can use something like: > > mkdir asterisk.git > cd asterisk.git > git svn init -T trunk -b branches -t tags http://svn.digium.com/svn/asterisk > git svn fetch > > to set up your own git clone of the Asterisk sources. Be prepared, > however - the first "git svn fetch" will take a long time to fetch > everything it needs. However, after the first one, subsequent "get svn > fetch"es only download new revisions. Thanks. How does it deal with svn:externals? What version of git do you use? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
