Jon Lawrence wrote:
Hi,
I've got 2 x100p's installed in my system.
Both execute the same incoming contexts as follows:
[inboundA]
include = dialjon
[inboundB]
include = dialjon|09:00-16:30|Mon-Fri|*|*
[dialjon]
exten = s,1,answer
exten = s,2,Dial(SIP/2000,15)
exten = s,3,Playback(noone)
exten =
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Monday, March 15, 2004 11:01 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] extensions problem (SIP)
Jon Lawrence wrote:
Hi,
I've got 2 x100p's installed in my system.
Both execute the same incoming
On Monday 15 March 2004 16:00, Olle E. Johansson wrote:
It depends on your SIP device. Asterisk places the call to your SIP device
regardless, since by SIP protocol design the UA is not a slave, it is
free. So the SIP ua must answer busy for Asterisk to understand that
you're busy. If not,
Jon Lawrence wrote:
Surely * should know if a phone is in use ? After all it initiated/took part
in the call in the first place ;)
Again, the SIP device is not a slave device. It could receive a call from
somewhere else and be busy without Asterisk having a clue. A lot of SIP UAs,
like Xten
On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
The incominglimit limits how many simultaneous calls a UA may place to
Asterisk.
I'm pretty sure that the incominglimit specifies how many calls that * can send to the
SIP device. If you set incominglimit=1 and then do a SIP
Walker Haddock wrote:
On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
The incominglimit limits how many simultaneous calls a UA may place to
Asterisk.
I'm pretty sure that the incominglimit specifies how many calls that * can send to the SIP device. If you set incominglimit=1
On Monday 15 March 2004 20:35, Walker Haddock wrote:
On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
The incominglimit limits how many simultaneous calls a UA may place to
Asterisk.
I'm pretty sure that the incominglimit specifies how many calls that * can
send to the
On Mon, Mar 15, 2004 at 08:56:17PM +, Jon Lawrence wrote:
The interface to my handytone is identical to a BT-102 so it may also work
with the handytone :). Where did you specify incominglimit=1 - is it in the
sip.conf for that UA ?
Yes, put it in the stanza for the devicd. As Olle just
On Mon, Mar 15, 2004 at 09:56:04PM +0100, Olle E. Johansson wrote:
Walker Haddock wrote:
On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
The incominglimit limits how many simultaneous calls a UA may place to
Asterisk.
I'm pretty sure that the incominglimit specifies
