try to set in your zapata.conf
overlapdial=yes
then in your asterisk cli
reload chan_zap.so
--
Marco Mouta
On Fri, Feb 13, 2009 at 9:21 AM, joek...@gmail.com wrote:
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context in
joek...@gmail.com schrieb:
Default FreePBX context,
[from-pstn]
The call seems to be going here
[ext-did-catchall]
So?
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = ext-did-post-custom
include = from-did-direct; MODIFICATOIN (PL) for findmefollow if
enabled, should be
Hi all,
I have a connect between a siemens hipath Asterisk system over PRI
The connection works perfectly I can call from the Hipath to an Asterisk
Extension.
I want to allow the hipath extensions to dial out over a SIP trunk on
asterisk but I keep getting The number you have dialed is not in
Which line of code is generating this log entry?
[Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing
[91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack
...because this appears to be where your problem lies.
joek...@gmail.com wrote:
Hi all,
I have a connect between a siemens hipath
I thing, you have bad routing configuration in extensions.conf. Send me
from-pstn context configuration.
turby
joek...@gmail.com napsal(a):
Hi all,
I have a connect between a siemens hipath Asterisk system over PRI
The connection works perfectly I can call from the Hipath to an
Asterisk
