In that case, I wonder how an SIP phone which is use SIP protocol to set up a call session can communicate with a SIP phone which is use H.323 if there is no module to convert two types of message Actually, we can use extension.conf to redirect from SIP channel to H.323 channel but we really need an converted module or converted function. Is that right?
@Jeremy: Thank for your answer so much, Regards, Thong Lam -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, October 05, 2006 2:16 PM To: asterisk-dev@lists.digium.com Subject: asterisk-dev Digest, Vol 27, Issue 16 Send asterisk-dev mailing list submissions to asterisk-dev@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-dev or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-dev digest..." Today's Topics: 1. Re: What's the best source for architectural understanding? (Jay R. Ashworth) 2. Re: Another bounty - app_reload (Matthew Rubenstein) 3. Re: app_reload? (Kristian Kielhofner) 4. Re: Re: app_reload? (Steve Edwards) 5. Re: app_reload? (Jeremy McNamara) 6. SIP to IAX (Thong Lam Hai) 7. Re: SIP to IAX (Jeremy McNamara) 8. Re: autoconf issues for FreeBSD (Luigi Rizzo) ---------------------------------------------------------------------- Message: 1 Date: Thu, 5 Oct 2006 00:03:23 -0400 From: "Jay R. Ashworth" <[EMAIL PROTECTED]> Subject: Re: [asterisk-dev] What's the best source for architectural understanding? To: asterisk-dev@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Wed, Oct 04, 2006 at 10:49:34PM -0500, Moises Silva wrote: > >So, one centralized dispatcher looks for both types of off-hooks on all > >channels? > No, for each telephony technology must exists a channel driver that > will "listen" or "expect" for incoming calls. Ah. > Once detects an incoming > call, will try to authenticate the call against its devices configured > in its configuration file (sip.conf, iax.conf, zapata.conf, > unicall.conf etc), and will start a new thread in the PBX (extensions > matching) for handling that new call. So, you have at least 1 listener > thread for each telephony technology. And 1 thread for each call > executing extensions in the dial plan. Aha. So there is actually an independent thread running for each active call, that you could inspect and log and manhandle. Good. Glad to hear that. Thanks for the clarification. Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & Associates The Things I Think '87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ------------------------------ Message: 2 Date: Thu, 05 Oct 2006 00:19:19 -0400 From: Matthew Rubenstein <[EMAIL PROTECTED]> Subject: Re: [asterisk-dev] Another bounty - app_reload To: Kristian Kielhofner <[EMAIL PROTECTED]> Cc: Asterisk-Dev <asterisk-dev@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain On Wed, 2006-10-04 at 17:31 -0700, [EMAIL PROTECTED] wrote: > Date: Wed, 04 Oct 2006 19:02:08 -0400 > From: Kristian Kielhofner <[EMAIL PROTECTED]> > Subject: Re: [asterisk-dev] Another bounty - app_reload > To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=iso-8859-1; format=flowed > > Andrew Kohlsmith wrote: > > On Wednesday 04 October 2006 15:47, Kristian Kielhofner wrote: > > > >> So, here is another bounty! app_reload should be able to > initiate a > >>reload from the dialplan. It should also be able to (optionally) > >>specify a single module to reload, i.e. chan_sip.so, etc. > > > > > > What's wrong with: > > > > exten => 735623,1,System(/usr/sbin/asterisk -rx "reload") > > exten => 735623,n,Hangup > > > > ?? > > > > -A. > > As I stated before: > > "Yes, obviously System("asterisk -rx reload") and other variations > would > work. But does using the System command to launch another Asterisk > process to connect to the already running Asterisk instance over a > UNIX > socket to issue a reload bother anyone else? It sure bothers me... > Wouldn't having a dialplan app to call ast_module_reload be sooo much > simpler?" Actually, what seems missing from the diaplan apps is one that just writes to the Manager API socket any passed string, and one to read from the Manager API socket any sent data. The read command would be even better if it could be connected to the API initially, and buffer data sent from the Manager, for parsing by regexp (and flushing). These apps would be especially useful in Perl Asterisk::AGI. That way dialplan and Perl (etc) could always offer a consistent API even with arbitrary added apps to the local installation. ManagerWrite(<command>, <delimited-data-string>) ManagerRead($[<channel-variable]) ManagerSubscribe($[channel-variable]) Right now it looks like the only way to do it is to write a Perl sub() in each Asterisk::AGI Perl script that opens a socket and reads from it. And maybe update the local install's pm. Much better if it's standardized as part of the API that everyone's AGIs (or diaplans) can call, depending on the installed Manager commands installed. > -- > Kristian Kielhofner > > -- (C) Matthew Rubenstein ------------------------------ Message: 3 Date: Thu, 05 Oct 2006 00:31:51 -0400 From: Kristian Kielhofner <[EMAIL PROTECTED]> Subject: [asterisk-dev] Re: app_reload? To: Jeremy McNamara <[EMAIL PROTECTED]> Cc: asterisk-dev@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Jeremy McNamara wrote: > Kristian Kielhofner wrote: > >> I'm amazed this hasn't been thought of before... >> >> Thanks again, you made it happen! > > > > I made app_reload report status to the calling party - It now streams > various standard prompts to the caller informing them if the reload > succeeded or not. > > If you hear 'thank you' that means the reload worked - There wasn't an > appropriate standard prompt that I could find, so I went with thank you. > > If you hear the beeperr tone (with no thank you) the reload failed - > meaning either there is another reload already happening or the passed > value didn't match a module that was loaded. > > > > Jeremy McNamara > > > > P.S. I am developing on the v1.2 version so the SVN trunk version may be > totally borken. > Jeremy, I need to send some prompts into Allison... I'll add these. What should she say? How about: Reload Successful Reload Failed Does app_reload set any channel variables based on status? I don't want this to turn into some kind of albatross for you, but I thought I'd ask :). -- Kristian Kielhofner ------------------------------ Message: 4 Date: Wed, 4 Oct 2006 21:41:56 -0700 (PDT) From: Steve Edwards <[EMAIL PROTECTED]> Subject: Re: [asterisk-dev] Re: app_reload? To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed 1) Reload 2) Successful 3) Failed 4) Extensions 5) Sip 6) Iax 7) Zap 8) ... On Thu, 5 Oct 2006, Kristian Kielhofner wrote: > Jeremy McNamara wrote: >> Kristian Kielhofner wrote: >> >>> I'm amazed this hasn't been thought of before... >>> >>> Thanks again, you made it happen! >> >> >> >> I made app_reload report status to the calling party - It now streams >> various standard prompts to the caller informing them if the reload >> succeeded or not. >> >> If you hear 'thank you' that means the reload worked - There wasn't an >> appropriate standard prompt that I could find, so I went with thank you. >> >> If you hear the beeperr tone (with no thank you) the reload failed - >> meaning either there is another reload already happening or the passed >> value didn't match a module that was loaded. >> >> >> >> Jeremy McNamara >> >> >> >> P.S. I am developing on the v1.2 version so the SVN trunk version may be >> totally borken. >> > > Jeremy, > > I need to send some prompts into Allison... I'll add these. What > should she say? How about: > > Reload Successful > Reload Failed > > Does app_reload set any channel variables based on status? I don't > want this to turn into some kind of albatross for you, but I thought I'd ask > :). > > -- > Kristian Kielhofner > _______________________________________________ > --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 > Thanks in advance, ------------------------------------------------------------------------ Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ------------------------------ Message: 5 Date: Thu, 05 Oct 2006 00:51:22 -0400 From: Jeremy McNamara <[EMAIL PROTECTED]> Subject: [asterisk-dev] Re: app_reload? To: Kristian Kielhofner <[EMAIL PROTECTED]> Cc: asterisk-dev@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Kristian Kielhofner wrote: > Does app_reload set any channel variables based on status? I don't > want this to turn into some kind of albatross for you, but I thought I'd > ask :). No, but it is trivial to set them - What were you thinkin? Jeremy McNamara ------------------------------ Message: 6 Date: Thu, 5 Oct 2006 14:09:44 +0700 From: "Thong Lam Hai" <[EMAIL PROTECTED]> Subject: [asterisk-dev] SIP to IAX To: <asterisk-dev@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Hi, Where is the code to change from SIP message to IAX message in Asterisk? How do 2 channels (such as channel_sip and channel_iax2) communicate with each other? Regards, Thong Lam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20061005/df78304b /attachment-0001.htm ------------------------------ Message: 7 Date: Thu, 05 Oct 2006 03:11:03 -0400 From: Jeremy McNamara <[EMAIL PROTECTED]> Subject: Re: [asterisk-dev] SIP to IAX To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Thong Lam Hai wrote: > Hi, > > > > Where is the code to change from SIP message to IAX message in Asterisk? There is no 'code' > How do 2 channels (such as channel_sip and channel_iax2) communicate > with each other? The Asterisk Dialplan (extensions.conf) Jeremy McNamara ------------------------------ Message: 8 Date: Thu, 5 Oct 2006 00:15:33 -0700 From: Luigi Rizzo <[EMAIL PROTECTED]> Subject: Re: [asterisk-dev] autoconf issues for FreeBSD To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Wed, Oct 04, 2006 at 08:37:45PM -0500, Kevin P. Fleming wrote: > ----- Luigi Rizzo <[EMAIL PROTECTED]> wrote: > > I am pretty sure that the code below (which was my patch at the time) > > is NOT a proper fix - i would expect autoconf to deal with platform > > issues > > without having us deal with them on each and every port. > > That fix probably won't work correctly any longer, since the change to use ASTCFLAGS and ASTLDFLAGS for the build system. as russel said, here i only patched CFLAGS and LDFLAGS which is what the configure script uses. AST* variables were already (more or less) set properly. > autoconf itself does nothing to increase platform compatibility; it is a tool that allows a package builder to increase compatibility. The benefits from autoconf are derived entirely from the logic that we put in the configure script, not autoconf itself. i am sure this has been discussed at length many times. I think the goal of autotools was to ease portability, both cross-platform and within a single platform with different packages installed. Maybe for the latter, autoconf still makes sense. But if you have to explicitly put in checks for what are platform defaults, the first goal is defeated. Besides, the "failures" are often subtle - it's not that configure fails completely, it just gives you a suboptimal build because it doesn't find 8 codecs and 4 libraries and so it doesn't build the dependent modules. And it does it so noisily (currently the output of asterisk's configure is 332 lines!) that it is easy to miss the failures while the text scrolls. Also the choice of using "ancient greek" (pre-1977 shell) as the language for expressing the logic in the configure script doesn't help correctness: e.g. without local variables, i think macros are not reentrant and things like saved_CFLAGS="${CFLAGS}" ... CFLAGS="${saved_CFLAGS}" could easily be trashed. > Once again (I'll agree with Russell here) it makes absolutely no sense for /usr/local to be the standard place for libraries to be installed, but that they are not in the search path for the compiler and linker. All this does is require every package that wants to build on FreeBSD to require special logic to add them to the search path, when the FreeBSD toolchain could easily have these paths embedded into the toolchain by default. How frustrating :-) Yes, frustrating. But blaming the platform (or autoconf as i did above, or my own ignorance as i often do privately) doesn't fix the build, so let's work on the latter :) cheers cheers: Command not found. /usr/local/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 End of asterisk-dev Digest, Vol 27, Issue 16 ******************************************** _______________________________________________ --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
