as second argument).
Aside from that: Are you sure that it is a wise idea to use symbolic
mailbox names (instead of only numeric)? You will not be able to enter a
mailbox on the phone when asked for it (i.e., VoiceMailMain() without an
argument).
--
Dr. Michael Neuhauser
on 1.2, but things are different there). Only use
this patch on a test system as it will generate massive amounts of
output and will considerably slow down call handling.
--
Dr. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH
or wait for the 1.2.13
release. Automon does work for me after the fix from the 1.2 branch was
applied.
--
Dr. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH sip:[EMAIL PROTECTED]
Vienna/Austria/Europe
On Tue, 2006-10-03 at 14:16 -0400, Clif Jones wrote:
Thanks for the response. Answers inline..
-Original Message-
From: Michael Neuhauser [EMAIL PROTECTED]
Sent: Oct 3, 2006 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, Clif
. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH sip:[EMAIL PROTECTED]
Vienna/Austria/Europe tel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at
, but im not sure. thanks.
That's not the way it works. Only the called phone (i.e., the ringing
one) is getting caller-id information (number/name), the calling one
(gxp in your example) does not.
--
Dr. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH
On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote:
michael,
at my real job, the phones display peoples names when calling out from
your phone. how is this done?
And the use Asterisk there? SIP phones? Legacy PBX? Cisco Call Manager?
--
Dr. Michael Neuhauser mailto:[EMAIL
--
Dr. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH sip:[EMAIL PROTECTED]
Vienna/Austria/Europetel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at/
___
--Bandwidth
+SendText).
--
Dr. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH sip:[EMAIL PROTECTED]
Vienna/Austria/Europetel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at
). Use ZAP/g1 (or whatever
your group is) - works for me on PRI (both as net and as cpe) -
depending on the behaviour of the other side ZAP/G1 could reduce the
likelihood of the same channel being used by both sides.
--
Dr. Michael Neuhauser mailto:[EMAIL PROTECTED
extension would be perfect.
Using this
exten = _*8.,1,DPickup(${EXTEN:1})
one can dial *8123 to pick up a call ringing at extension 123. Works for
me with Asterisk 1.2.11 and bristuff 0.3.0-1s
--
Dr. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH
. Michael Neuhauser mailto:[EMAIL PROTECTED]
Firmix Software GmbH sip:[EMAIL PROTECTED]
Vienna/Austria/Europe tel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at
lines added) that fixes this. The same bug is still present in
Asterisk 1.2.5
Regards,
Mike
--
Dr. Michael Neuhauserphone: +43 1 789 08 49 - 30
Firmix Software GmbH fax: +43 1 789 08 49 - 55
Vienna/Austria/Europe email: [EMAIL PROTECTED
, and in addition to CLI[PR]) before the call is
answered?
The USER-USER information element of Q.931 might do the trick, but
support for that is still disabled in Asterisk 1.2.5 (look for
SUPPORT_USERUSER in chan_zap.c). Maybe it works with one of the
dedicated isdn channel modules.
--
Dr. Michael Neuhauser
be equivalent (and both correct) if the
definition in transcap.h followed best practices to avoid problems with
operator precedence and looked like this:
#define IS_DIGITAL(cap)\
((cap) AST_TRANS_CAP_DIGITAL ? 1 : 0)
Regards,
Mike
--
Dr. Michael Neuhauser
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