[asterisk-dev] UNSUBSCRIBE
Brien Hamrick
brienhamrick at sbcglobal.net
Mon Jun 1 12:35:33 CDT 2009
I don't think Christ would have made that mistake, it's likely Freddy on his
own :-)
-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Steve Howes
Sent: Monday, June 01, 2009 9:10 AM
To: Freddy Mangum; Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] UNSUBSCRIBE
Christ you even forwarded the instructions you ignored..
On 1 Jun 2009, at 16:37, Freddy Mangum wrote:
>
> Cheers,
>
> Freddy Mangum
> fjmangum at gmail.com
>
>
>
> On Jun 1, 2009, at 8:25 AM, asterisk-dev-request at lists.digium.com
> wrote:
>
>> Send asterisk-dev mailing list submissions to
>> asterisk-dev at 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
>> asterisk-dev-request at lists.digium.com
>>
>> You can reach the person managing the list at
>> asterisk-dev-owner at lists.digium.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of asterisk-dev digest..."
>>
>>
>> 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
>>>
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL:
http://lists.digium.com/pipermail/asterisk-dev/attachments/20090531/219b2ba6
/attachment-0001.htm
>>
>> ------------------------------
>>
>> 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
>>>
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL:
http://lists.digium.com/pipermail/asterisk-dev/attachments/20090601/bafcccd9
/attachment-0001.htm
>>
>> ------------------------------
>>
>> 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
>>>
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL:
http://lists.digium.com/pipermail/asterisk-dev/attachments/20090601/8200941e
/attachment.htm
>>
>> ------------------------------
>>
>> _______________________________________________
>> --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
>>
>> End of asterisk-dev Digest, Vol 59, Issue 1
>> *******************************************
>
>
> _______________________________________________
> --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
_______________________________________________
--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
More information about the asterisk-dev
mailing list