I don't know if I understand you correctly but you could place a Goto or a
Hangup there:
exten => 99,1,Gotoif(?2:4)
exten => 99,2,Meetme(100)
exten => 99,3,Goto or Hangup
exten => 99,4,Meetme(100|options)
> -Original Message-
> From: nik600 [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, J
Hi all,
I am using Asterisk 1.2.10 on Debian Sarge and currently I am rewriting my
extensions.conf with ael.
The replacement of the following part makes me mad:
[set-language]
exten => _X./_0031.,1,Set(incoming_call=1|lang=nl)
exten => _X./_0031.,2,Goto(incoming,${EXTEN},1)
exten => _X./_0049.,1
No, the Gigaset is the only WLAN phone I tested so long, so I can not
compare it to the other phones you mentioned.
-Original Message-
From: Olivier [mailto:[EMAIL PROTECTED]
Sent: Friday, November 24, 2006 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [
I bought the phone in Germany. Except another wlan phone from Siemens which
was not available any more, I did not find any alternatives to it.
-Original Message-
From: Olivier [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 22, 2006 8:44 AM
To: Asterisk Users Mailing List - Non-Commerc
Hi all,
when calling from the PSTN with "Match As You Go" Dialing (lift the handset
before start to dial) over a zap channel Asterisk simply takes the first
extension digit and tries to match it. Because no valid one-digit-extension
exists in my dialplan, matching fails and Asterisk says that a in
Hi all,
I have problems receiving calls from PSTN with an Wildcard T207P.
All internal SIP devices have a 3 digit extension, e.g. 873.
When I call the extension from the PSTN this way everything works fine:
1. enter the number on the phone
2. lift off the handset
But when I call it that way A
Hi all,
for the user with extension 191 I get the following error message:
ERROR[10131]: chan_sip.c:10831 handle_request_subscribe: Got SUBSCRIBE for
extensions without hint. Please add hint to 191 in context internal
I don't know where to add something because the entries in sip.conf look the
s
Hi all,
I've read a lot of problems with faxing over asterisk. Most of them referred
to Fax over Internet, but I want to connect analog and ISDN fax devices to
asterisk to send and receive faxes over PRI:
+-+ +--+++
| | | || ISDN Fax |
| PRI
Hi !
I am trying to transfer calls between internal SIP softclients, but it does
not work. Every time I press a key on the softclient, the CLI shows the
following output:
Attempting native bridge of SIP/456-9ee0 and SIP/173-f586
This is my extensions.conf:
[macro-voicemail]
exten => s,1,Dial(${
