I have been told that the combination of call waiting, * and FXO does and will not work because "Asterisk is a PBX". I guess I'd like to hear if this is a hard and fast "no this will not work and here's why", or that this currently doesn't work but with some coding might work. I'd like to have the option to be able to continue using call waiting with an FXO line (and I know I'm not alone). I know if I switched to a SIP based connection instead of the FXO this would work, but I currently like my unlimited plan with Vonage. Would anyone like to enlighten me? I have done numerous searches and I've included a few postings that were mostly not answered. http://lists.digium.com/pipermail/asterisk-users/2004-May/046855.html http://www.vovida.org/pipermail/mgcp/2001-May/000571.html Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040728/30de3b01/attachment.htm
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Wednesday 28 July 2004 11:38 pm, mike jennings wrote:> I have been told that the combination of call waiting, * and FXO does and > will not work because "Asterisk is a PBX". I guess I'd like to hear if > this is a hard and fast "no this will not work and here's why", or that > this currently doesn't work but with some coding might work.The way I see it call waiting is kinda crazy with a PBX. A PBX is really there to handle a situation with a number of lines and a number of stations (phones). A call comes in, it's then routed to an internal phone. If someone calls the same number they are really supposed to end up in a hunt group, and come in on the next free line. Instead it interrupts the previous call who probably have nothing to do with the new caller. Call waiting is a substitute for having more than one line. This is highly unlikely to occur in a PBX environment, which for no other reason than price, is usually not available in a single line home. Technically, with a card (if it exisits) that can handle the hook flash, there's probably not any reason why one could not have call waiting play confusion with people and interrupt them. It just does not make any practical sense to do so. SIP phones are often used without a PBX so there it makes sense to have it available. - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBCHmHljK16xgETzkRAp8tAJ0SCaIlC1LMCq5hqZjNpQ6cFEx6ewCfX0oJ hFeN6JmKVN4pxUaODX27aLs=tIao -----END PGP SIGNATURE-----
Not in any way a good solution, but what I've done is create an extension
that flashs the line, and then returns the call to my sip phone. For example:
[app-flash]
exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})
[macro-test]
exten => s,1,Answer
exten => s,3,Flash
exten => s,3,Dial(SIP/${ARG2},30,t)
exten => s,4,Dial(SIP/${ARG1},30,t)
exten => s,t,Hangup
exten => s,i,Hangup
exten => s,h,Hangup
Then if you're on a call through the Zap line, and transfer the call to
*4xxxx, it will flash the line and return it to xxxx SIP extension. I've
been trying to get it to auto-detect the SIP extension to return it to, but
callerid is different depending on if the call is incoming or outgoing through
the Zap.
Again, not good.. but works in a home environment. I think we'll need
in-call triggers to do anything better.
Ben Wern
---------- Original Message -----------
From: "mike jennings" <mike.jennings@charter.net>
To: <asterisk-users@lists.digium.com>
Sent: Wed, 28 Jul 2004 22:38:41 -0500
Subject: [Asterisk-Users] call waiting, * and FXO
> I have been told that the combination of call waiting, * and FXO does and
will not work because ?Asterisk is a PBX?.? I guess I?d like to hear if this is
a hard and fast ?no this will not work and here?s why?, or that this currently
doesn?t work but with some coding might work.
>
?
> I?d like to have the option to be able to continue using call waiting with
an FXO line (and I know I?m not alone).? I know if I switched to a SIP based
connection instead of the FXO this would work, but I currently like my unlimited
plan with Vonage.?
>
?
> Would anyone like to enlighten me?
>
?
> I have done numerous searches and I?ve included a few postings that were
mostly not answered.
>
?
> http://lists.digium.com/pipermail/asterisk-users/2004-May/046855.html
> http://www.vovida.org/pipermail/mgcp/2001-May/000571.html
>
?
> Thanks
------- End of Original Message -------
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