[asterisk-dev] UNSUBSCRIBE
Freddy Mangum
fjmangum at gmail.com
Mon Jun 1 10:37:57 CDT 2009
Cheers,
Freddy Mangum
fjmangum at gmail.com
On Jun 1, 2009, at 8:25 AM, asterisk-dev-request at lists.digium.com wrote:
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> Today's Topics:
>
> 1. Re: wanna be asterisk-dev (Valter Nogueira)
> 2. Re: wanna be asterisk-dev (Kevin P. Fleming)
> 3. [Code Review] Multicast RTP Paging Support (Joshua Colp)
> 4. Re: wanna be asterisk-dev (Valter Nogueira)
> 5. Re: wanna be asterisk-dev (Tzafrir Cohen)
> 6. Re: wanna be asterisk-dev (Valter Nogueira)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sun, 31 May 2009 19:42:48 -0300
> From: Valter Nogueira <vgnogueira at gmail.com>
> Subject: Re: [asterisk-dev] wanna be asterisk-dev
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID:
> <48aca8a40905311542x52191668t57f8ce9a83ce1d45 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Nir,
>
> thanks for your comments.
>
> My outbound campaigns are strutured as a set of CONTEXTS - including
> IVR
> options.
>
> Each action in the campaign is transformed into a CONTEXT. My end-
> users
> program each campaingn as follow
>
> ACTION: 010_START NEXT_ACTION:
> 020_PLAY_WELCOME
> ACTION: 020_PLAY_WELCOME NEXT_ACTION: 030_IVR
> ACTION: 030_IVR NEXT_ACTION:
> NONE 1_ACTION: ...., #_ACTION: ....
>
> Every action I transform into a CONTEXT that executes some asterisk
> app and
> finally branchs to the next one - some what like this:
>
> [010_START]
>
> exten => s,1,agi(callprogress, 'success')
> exten => s,n,goto(020_PLAYWELCOME, s,1)
> exten => s,n,hangup()
>
> exten => failed,1,agi(callprogress,'failed')
> exten => failed,n,hangup()
>
> [020_PLAYWELCOME]
>
> exten => s,1,agi(callprogress, 'playback', filename)
> exten => s,n,playbakc(filename)
> exten => s,n,goto(030_IVR,s,1)
>
> [030_IVR]
>
> exten => s,1,agi(callprogress,'ivr')
> exten => s,n,waitexten()
>
> exten => 1,1,goto(.......)
>
> When I originate the call using a ZAP channel and I direct it to
> extension
> 010_START, s, 1 - if originate fails it leads automatically to
> 010_START,
> failed, 1
>
> This way I was able to produce manually elaborated campaigns using
> just
> extension.conf and a python script that generated callfiles.
>
> Now I am going a step further making all that things automatically
>
> Thanks,
>
> Valter
>
>
> 2009/5/31 Nir Simionovich <nir.simionovich at gmail.com>
>
>> Hi Valter,
>>
>> Welcome to the DEV list, please find my comments below:
>>
>> Valter Nogueira wrote:
>>> Hi people,
>>>
>>> I am a wanna be asterisk-dev, but the learning curve is somewhat
>>> hard.
>>>
>>
>> As a newbie developer myself, I agree that the learning curve is
>> steep and hard. The best way to learn is to read various code
>> snippets
>> and try to extrapolate what they do. That's what I did when I started
>> working on adding Hebrew to app_voicemail.c and say.c
>>
>>> Do you have any clues such "IDE", debugging tools and what ever
>>> could
>> help?
>>>
>>
>> Well, I'm using a mixture of Eclipse with the CDT plugins and VIM.
>> Debugging is always done with GDB - other than that, just follow the
>> Digium coding guidelines.
>>
>>> Should I start on 1.4 or 1.6 version - I am using 1.4 as user so I
>>> think
>>> it is most apropriate stick with it as dev.
>>>
>>
>> Actually, the best choice will be SVN. If you write a new feature,
>> it
>> will only be evaluated on the SVN. If you fix a bug for 1.6 or 1.4,
>> that
>> can be done directly on the version, however, still needs to be
>> ported
>> to the SVN version. Just like you, I started working on the 1.4
>> version,
>> quickly to realize that I need to do my work twice each time - so I
>> migrated to SVN version.
>>
>>> I am developing an Asterisk based dialer. My first version write
>>> down an
>>> extension.conf - so I reload it every time config changes.
>>>
>>> So, I thought: What if I create extensions on the fly: "DIALPLAN ADD
>>> EXTENSION 1111,1,COMMAND INTO MYCONTEXT"
>>>
>>> It sounded great, but just worked if MYCONTEXT is already present
>>> (what
>>> is not always the case).
>>>
>>> I thought again (what is not usual): Source code can answer it to
>>> me.
>>>
>>> Now the hard part where in the damn giant source code DIAL PLAN ADD
>>> EXTENSION is handled.
>>>
>>> Few greps after I found it -> pbx_config.c - where I found 2
>>> function -
>>> one marked as deprecated
>>>
>>> How should I debug asterisk to get here and understand what is
>>> happening?
>>>
>>> Should asterisk create a new dialplan context when adding an
>>> extension?
>>>
>>> Is there some other way to dynamically add or remove CONTEXTS?
>>>
>>> Do you have any tips for me.
>>>
>>
>> As far as I know, these functions are now available via the Asterisk
>> manager. In any case, I'm not sure you are going in the right
>> direction,
>> as I can't see a relation between a dialer and auto-generating
>> extensions.conf. My only logic here would be that you are
>> originating a
>> call out of Asterisk, then, creating a special dialplan for that
>> call,
>> re-directing the generated the call to the newly created dialpan.
>> In any
>> case, it would suggest that you're over complicating things (at
>> least in
>> my book - literally).
>>
>> Have a safe journey in learning the inner workings of Asterisk's
>> source code. If you drink coffee (or alcohol for that matter), this
>> would be a good point to pour yourself a pint of your finest beer,
>> gulp
>> it down and use the code.
>>
>>> Thank you all,
>>>
>>> Valter
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> --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
>>
>>
>> --
>> Kind Regards,
>> Nir Simionovich
>> Asterisk Community Founder and Maintainer - Israel
>>
>> (e) nir.simionovich at gmail.com
>> (w) http://www.simionovich.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
>>
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> ------------------------------
>
> Message: 2
> Date: Mon, 01 Jun 2009 06:56:23 -0500
> From: "Kevin P. Fleming" <kpfleming at digium.com>
> Subject: Re: [asterisk-dev] wanna be asterisk-dev
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <4A23C1E7.50300 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Valter Nogueira wrote:
>
>> This way I was able to produce manually elaborated campaigns using
>> just
>> extension.conf and a python script that generated callfiles.
>>
>> Now I am going a step further making all that things automatically
>
> As a simple step, you can use #exec from within extensions.conf to run
> another Python script to generate all the needed contexts; you still
> have to 'reload' to get them into memory, but you don't have to edit
> any
> text files or duplicate the information.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kpfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 01 Jun 2009 13:07:07 -0000
> From: "Joshua Colp" <jcolp at digium.com>
> Subject: [asterisk-dev] [Code Review] Multicast RTP Paging Support
> To: "Joshua Colp" <jcolp at digium.com>, "Asterisk Developers"
> <asterisk-dev at lists.digium.com>
> Message-ID: <20090601130707.11388.22442 at hotblack.digium.internal>
> Content-Type: text/plain; charset="utf-8"
>
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> http://reviewboard.digium.com/r/270/
> -----------------------------------------------------------
>
> Review request for Asterisk Developers.
>
>
> Summary
> -------
>
> This patch implements two new modules: res_rtp_multicast and
> chan_multicast_rtp. The resource module is an RTP engine which can
> be used by any developer to send multicast RTP. The channel driver
> uses the RTP engine and configures it based on input given from the
> user in the Dial line. Any audio sent to the RTP engine is broadcast
> out.
>
>
> This addresses bug 11797.
> https://issues.asterisk.org/view.php?id=11797
>
>
> Diffs
> -----
>
> /trunk/channels/chan_multicast_rtp.c PRE-CREATION
> /trunk/res/res_rtp_multicast.c PRE-CREATION
>
> Diff: http://reviewboard.digium.com/r/270/diff
>
>
> Testing
> -------
>
> Confirmed that audio is sent out as expected by dialing using the
> channel driver in the dialplan.
>
>
> Thanks,
>
> Joshua
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Mon, 1 Jun 2009 11:22:37 -0300
> From: Valter Nogueira <vgnogueira at gmail.com>
> Subject: Re: [asterisk-dev] wanna be asterisk-dev
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID:
> <48aca8a40906010722y63920f20r1c7158cfae57ee7f at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Kevin,
>
> I know it is a little off-topic - I will try #exec - thanks.
>
> Does it works for others conf files such meete, iax and soon? Or
> should I
> try Asterisk Realtime?
>
> I am thinking in moving my context based workflow to a fastagi one -
> this
> would led me to a single context entry and a state machine - with no
> reloads
> at all.
>
> Valter
>
>
>
>
> 2009/6/1 Kevin P. Fleming <kpfleming at digium.com>
>
>> Valter Nogueira wrote:
>>
>>> This way I was able to produce manually elaborated campaigns using
>>> just
>>> extension.conf and a python script that generated callfiles.
>>>
>>> Now I am going a step further making all that things automatically
>>
>> As a simple step, you can use #exec from within extensions.conf to
>> run
>> another Python script to generate all the needed contexts; you still
>> have to 'reload' to get them into memory, but you don't have to
>> edit any
>> text files or duplicate the information.
>>
>> --
>> Kevin P. Fleming
>> Digium, Inc. | Director of Software Technologies
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> skype: kpfleming | jabber: kpfleming at digium.com
>> Check us out at www.digium.com & www.asterisk.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
>>
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> ------------------------------
>
> Message: 5
> Date: Mon, 1 Jun 2009 17:50:33 +0300
> From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
> Subject: Re: [asterisk-dev] wanna be asterisk-dev
> To: asterisk-dev at lists.digium.com
> Message-ID: <20090601145033.GT3227 at xorcom.com>
> Content-Type: text/plain; charset=us-ascii
>
> On Mon, Jun 01, 2009 at 11:22:37AM -0300, Valter Nogueira wrote:
>> Kevin,
>>
>> I know it is a little off-topic - I will try #exec - thanks.
>>
>> Does it works for others conf files such meete, iax and so on?
>
> Yes, works for just about any Asterisk config file. Basically:
> anywhere
> '#include' will work (assuming you enabled using #exec in
> asterisk.conf). As the '#' hist, #include and #exec are preprocessing
> directives that are essentially procssed before the configuration is
> read just like the C pre-processor.
>
> But no, there's aren't any #if* or #define directives.
>
> OTOH, we have something called templaets. And like the thing in C++,
> it
> works in a more sematic level and supports (some sort of) inheritance.
> Unlike the thing from C++ and #exec, it's not turing-complete.
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.cohen at xorcom.com
> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>
>
>
> ------------------------------
>
> Message: 6
> Date: Mon, 1 Jun 2009 12:25:02 -0300
> From: Valter Nogueira <vgnogueira at gmail.com>
> Subject: Re: [asterisk-dev] wanna be asterisk-dev
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID:
> <48aca8a40906010825u1c8546f6n6669aadb0d552d8e at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> which source file do the job - I mean preprocesses conf files?
>
> By the way, is there any way to create/delete context on-fly?
>
> Valter
>
> 2009/6/1 Tzafrir Cohen <tzafrir.cohen at xorcom.com>
>
>> On Mon, Jun 01, 2009 at 11:22:37AM -0300, Valter Nogueira wrote:
>>> Kevin,
>>>
>>> I know it is a little off-topic - I will try #exec - thanks.
>>>
>>> Does it works for others conf files such meete, iax and so on?
>>
>> Yes, works for just about any Asterisk config file. Basically:
>> anywhere
>> '#include' will work (assuming you enabled using #exec in
>> asterisk.conf). As the '#' hist, #include and #exec are preprocessing
>> directives that are essentially procssed before the configuration is
>> read just like the C pre-processor.
>>
>> But no, there's aren't any #if* or #define directives.
>>
>> OTOH, we have something called templaets. And like the thing in C+
>> +, it
>> works in a more sematic level and supports (some sort of)
>> inheritance.
>> Unlike the thing from C++ and #exec, it's not turing-complete.
>>
>> --
>> Tzafrir Cohen
>> icq#16849755 jabber:tzafrir.cohen at xorcom.com<jabber%3Atzafrir.cohen at xorcom.com
>> >
>> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
>> http://www.xorcom.com iax:guest at local.xorcom.com/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
>>
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