I don't think this will solve your issue but I would say remove "r"
option in dial command I had same issue with iax and sip phone and I
solved with that r option
Hope it will help you.
--
Sent from my iPhone
On Apr 30, 2011, at 1:15 PM, "Roy Kidder" <[email protected]> wrote:
Hello,
I've got a problem with something I'm doing and can't seem to figure
it
out. I've tried different suggestions I've found on voip-info.org as
well
as other sites but nothing I do seems to work.
I've got an older Digium TDM400P. The FXO daughter card is connected
to my
POTS line and the FXS daughter card is connected to a TDM phone. I
also
have multiple SIP extensions. My desire is to ring all the internal
extensions (the TDM and SIP extensions) on an inbound call and send
the
call to whichever extension picks up first.
This seems to be working just fine if the extension that picks up is
one
of the SIP phones. On the other hand, if the extension that picks up
is
the one off the FXS port, then the SIP phones continue to ring and the
dial plan continues to execute even though the caller on the FXO
port has
been connected to the phone on the FXS port.
Inbound calls are sent to extension 3100, which looks like this:
exten => 3100,1,Dial(SIP/3105&SIP/3106&SIP/3108&dahdi/1,20,tr)
exten => 3100,n,Voicemail(3100)
exten => 3100,n(end),Hangup()
Like I said, if I pick up on one of the SIP extensions, it seems to do
exactly as I expect. If I pick up on dahdi/1, however, the SIP phones
continue to ring and the FXO and FXS ports are connected and passed
into
voicemail.
I'm running on Debian Squeeze/6.0.1 and am running the stock Asterisk
1.6.2.9-2+squeeze2 package.
If anyone has some suggestions, I'd be happy to hear them.
Thanks!
Roy
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users