asterisk users - Oct 2007 - Mask Initial Processing with Ring Back Tone

I need to process a number of lines of code in the dialplan before answering a
call.  Can standard ring back tones be played to the caller while this is
happening prior to answering the call.  Which commands would facilitate this?

Thanks in Advance,

Vic

On 10/11/07, Victor <voicecomputing at gmail.com>
wrote:
>
> I need to process a number of lines of code in the dialplan before
> answering a
> call.  Can standard ring back tones be played to the caller while this is
> happening prior to answering the call.  Which commands would facilitate
> this?

You start sending ringback with Ringing()

You answer with Answer()

What you do in between is up to you.  Many people use something like Wait(2)
to give a "comfort ring", since PRI-connected incoming calls can often
be
set up nearly instantaneously.  You'd want to limit the time obviously, and
have proper exception handling in case whatever you're doing between
Ringing() and Answer() fails.

-- 
j.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20071011/f934a8c2/attachment.htm

Victor wrote:
> I need to process a number of lines of code in the dialplan before
answering a
> call.  Can standard ring back tones be played to the caller while this is
> happening prior to answering the call.  Which commands would facilitate
this?
I strongly doubt those lines are going to take up much time.

You can use Playtones to play specific inband tones.

Just dont answer it till the processing is done.  No debate is needed  
for this.  I do this millions of times per month.

/b

On Oct 11, 2007, at 2:56 PM, "Eric \"ManxPower\" Wieling"
<eric at fnords.org
 > wrote:
> Victor wrote:
>> I need to process a number of lines of code in the dialplan before  
>> answering a
>> call.  Can standard ring back tones be played to the caller while  
>> this is
>> happening prior to answering the call.  Which commands would  
>> facilitate this?
>
> I strongly doubt those lines are going to take up much time.
>
> You can use Playtones to play specific inband tones.
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

Brian West wrote:
> Just dont answer it till the processing is done.  No debate is needed  
> for this.  I do this millions of times per month.
>   
Yes, this is one of those things too simple to be obvious.  Like Brian 
said, just do your processing and THEN Answer()  -- Generally, the 
caller will get ringback already, from their telco.

Am I correct in understanding that if the call comes in g729 and it is ended in
g729 ( by the provider ) , asterisk does only bridging, therefore using very few
CPU ressources ?

Am I correct in understanding that this "bridging" means that calls (
rtp ) pass from one provider to another, therefore using low bandwith?

Thank you....

A.


Helping businesses save money worldwide
www.sunapemobil.ca
Numar de acces in .ro : +40318107430 (se taxeaza la pretul unui numar 031) 
Romania 3.5 c/min (USD)
Moldova 10c/min
       
---------------------------------
Looking for a deal? Find great prices on flights and hotels with Yahoo!
FareChase.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20071011/1ba5fc47/attachment.htm

That is brought to you by the sip reinvite, in short yes, unless you set 
canreinvite = no to either side of that.

Apa Minerala wrote:
> Am I correct in understanding that if the call comes in g729 and it is 
> ended in g729 ( by the provider ) , asterisk does only bridging, 
> therefore using very few CPU ressources ?
>
> Am I correct in understanding that this "bridging" means that
calls (
> rtp ) pass from one provider to another, therefore using low bandwith?
>
> Thank you....
>
> A.
>
>
> Helping businesses save money worldwide
> www.sunapemobil.ca
> Numar de acces in .ro : +40318107430 (se taxeaza la pretul unui numar 
> 031)
> Romania 3.5 c/min (USD)
> Moldova 10c/min
>
> ------------------------------------------------------------------------
> Looking for a deal? Find great prices on flights and hotels 
>
<http://us.rd.yahoo.com/evt=47094/*http://farechase.yahoo.com/;_ylc=X3oDMTFicDJoNDllBF9TAzk3NDA3NTg5BHBvcwMxMwRzZWMDZ3JvdXBzBHNsawNlbWFpbC1uY20->
> with Yahoo! FareChase.
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP

I am *really* sorry about hijacking this thread, but the only way I can 
post to the -user list is by replying to another thread. (btw, this is 
getting really annoying. Please, Digium, sort the filters out!)

I installed my super-duper new TE412P card today, without remembering to 
check the settings for T1/E1.

As the server is now a hundred miles away, is there

a) Any way of checking what setting is in place
b) Changing that setting

without having to physically remove the card and see ?

Julian.

Eric "ManxPower" Wieling wrote:
> Julian Lyndon-Smith wrote:
>> Hijacking a thread again - the only way I can post to the -user list is
>> by replying to another thread. (btw, this is getting really annoying. 
>> Please, please, please, Digium, sort the filters out!)
> 
> You seem to be subscribed to the list as "asterisk at dotr.com". 
Is that
> the e-mail address you are using when you are trying to start a new 
it is indeed ;)

Julian.
> thread.  If the mailing list allowed messages from non-subscriber e-mail 
> address the list would be destroyed by spam.
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ______________________________________________________________________
> This email for dotr.com has been scanned by MessageLabs
> ______________________________________________________________________
> 
>

The cards ship configured for T1.  If you didn?t change the jumpers, it is
set for T1.

If it is set for T1 and you really want an E1 and you configure your
zapata.conf as you would for an E1, you will get an error around channel 25,
which tells you that you forgot to change the jumpers, and you have to call
the guy on site and ask him (again) to close the jumpers on the card....
This has never happened to me of course, but it happens regularly to this
guy that I know..



-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]En nombre de Julian
Lyndon-Smith
Enviado el: viernes, 12 de octubre de 2007 0:08
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] really sorry about this - E1 vs T1


I am *really* sorry about hijacking this thread, but the only way I can
post to the -user list is by replying to another thread. (btw, this is
getting really annoying. Please, Digium, sort the filters out!)

I installed my super-duper new TE412P card today, without remembering to
check the settings for T1/E1.

As the server is now a hundred miles away, is there

a) Any way of checking what setting is in place
b) Changing that setting

without having to physically remove the card and see ?

Julian.


_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users