[Asterisk-Users] IAX termination services

Mike Fedyk mfedyk at mikefedyk.com
Wed Jan 4 23:28:42 MST 2006


If you are not big enough to have your own domain, then you don't need a 
disclaimer.

This e-mail transmission may contain information that is non-proprietary,
unprivileged and/or non-confidential and is intended exclusively to provide a clue
to Jason D. Wolfe. Any use, copying, retention or disclosure by any
person other than the intended recipient or the intended recipient's
designees is strictly allowed. If you are not the intended recipient or
their designee, please distribute immediately so that people who try to wrap
a contract around an insecure medium as a means of security will wisen up.



Jason D. Wolfe wrote:

>I'm a newbie to Asterisk and telecom, and I I learned the hard way that
>analog POTS lines cause asterisk to start your dialplan as soon as the
>outbound starts ringing... that's why I was a little nervous about whether
>or not I may have the same problem using an IAX termination service.  As it
>turns out, it works perfectly, as they do provide 'answer supervision' (like
>all digital lines).
>
>as well, I'm not going to erase my disclaimer below every time I send an
>email to a listserv.  It does say 'person(s) to whom it is addressed', which
>keeps it from being completely senseless! :) and, I do NOT work for
>Bellsouth, they are my ISP...
>
>Jason Wolfe
>jasonwolfe at bellsouth.net
>
>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Jean-Michel
>Hiver
>Sent: Tuesday, January 03, 2006 1:01 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] IAX termination services
>
>
>Jason D. Wolfe a écrit :
>
>  
>
>>Hello,
>>
>>If I use an IAX termination service to connect outgoing VoIP calls to a
>>    
>>
>PSTN
>  
>
>>will I have answer supervision so that my script won't initiate too early?
>>
>>
>>    
>>
>I'm not sure to understand you. If you don't use Answer() before you use
>Dial(), asterisk won't answer until the dialed party does so.
>
>Cheers,
>Jean-Michel.
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>  
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060104/47c0e6ba/attachment.htm


More information about the asterisk-users mailing list