asterisk users - May 2007 - xten will not send tones to * and i from sip phone

hi there!

I have a couple phones connected to a sipura ata and
if I go into *- IVR, I press options on the regular
phones and it all works fine and dandy.

then I connect an xten softphone, a new extension in
my dialplan, I dial the ivr, * asks me to dial
something to go through it, I press keys on xten, but
nothing happens, * just times out through as if I did
not press anything!

is there some sort of configuration out there to tell
the xten softphone to work as expected? thanks!

Then another problem!

I used the i extension, plus _X and _X. to make sure I
catch everything that is not propperly dialed.

If I take the regular phones that are connected
through the sipura ata, then dial 'exten =>
700,1,Goto(default,s,1)' so that I get the asking for
an extension to reach, I dial a wrong number and
walla, its caight by one of my magic numbers!

BUT, if I pickup the same phone, and just dial the
same wrong number? I just get a busy signal! and there
is nothing registered at the CLI even though I added
DEBIG to the configuration! :s

What can I do to make sure I always send an error
sound and never again a busy signal?


thanks!






 
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>From: pedro noticioso <cucnews@yahoo.com>
>Date: Fri, 18 May 2007 20:36:59 -0700 (PDT)
>
>hi there!
>
>I have a couple phones connected to a sipura ata and
>if I go into *- IVR, I press options on the regular
>phones and it all works fine and dandy.
>
>then I connect an xten softphone, a new extension in
>my dialplan, I dial the ivr, * asks me to dial
>something to go through it, I press keys on xten, but
>nothing happens, * just times out through as if I did
>not press anything!
>
>is there some sort of configuration out there to tell
>the xten softphone to work as expected? thanks!
Seems to be a mismatch in dtmfmode between Asterisk and Xten.  You may try 
dial ***7469 in Xten to bring up a "magic" menu.  force_send_inband is
enabled by default - not sure if this has any bearing, though, because my 
Xlite works with my Asterisks by default.
>Then another problem!
>
>I used the i extension, plus _X and _X. to make sure I
>catch everything that is not propperly dialed.
>
>If I take the regular phones that are connected
>through the sipura ata, then dial 'exten =>
>700,1,Goto(default,s,1)' so that I get the asking for
>an extension to reach, I dial a wrong number and
>walla, its caight by one of my magic numbers!
>
>BUT, if I pickup the same phone, and just dial the
>same wrong number? I just get a busy signal! and there
>is nothing registered at the CLI even though I added
>DEBIG to the configuration! :s
Are you talking about the same context? (Specifically, does Sipura use 
[default]?)  Snippets of sip.conf and extensions.conf would be helpful if 
you are not sure what to look.

Yuan Liu
>What can I do to make sure I always send an error
>sound and never again a busy signal?
>
>
>thanks!