Re: [asterisk-users] Filtering on from caller id

2011-03-25 Thread Peter den Hartog
I/1,w,5551212) > > Exten => _X.,n,hangup > > > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Peter den Hartog > *Sent:* Friday, March 25, 2011 3:15 AM

Re: [asterisk-users] Filtering on from caller id

2011-03-25 Thread Peter den Hartog
27;s > -Original Message- > From: Peter den Hartog > Sender: asterisk-users-boun...@lists.digium.com > Date: Fri, 25 Mar 2011 09:14:45 > To: Asterisk Users Mailing List - Non-Commercial Discussion< > asterisk-users@lists.digium.com> > Reply-To: Asterisk

Re: [asterisk-users] Filtering on from caller id

2011-03-25 Thread Peter den Hartog
t; Exten => _X,1,noop(everybody but 103 dials) > > Exten => _X./103,n,hangup > > Exten => _X.,n,Dial(DAHDI/1,w,5551212) > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.c

Re: [asterisk-users] Filtering on from caller id

2011-03-24 Thread Peter den Hartog
hu, Mar 24, 2011 at 5:14 PM, A J Stiles wrote: > On Thursday 24 Mar 2011, Peter den Hartog wrote: > > > I would like to use the from caller id, to allow calls yes or no. > > 101, and 111 should be allowed to use the Trunk, the rest of the phones > are > > not. > > &

[asterisk-users] Filtering on from caller id

2011-03-24 Thread Peter den Hartog
Hi, I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Thanks, Peter -- __

Re: [asterisk-users] problem with crashing Asterisk 1.8

2011-03-11 Thread Peter den Hartog
Hmm disabled Woomera and everything seems stable. Strange! On Thu, Mar 10, 2011 at 11:46 AM, Peter den Hartog wrote: > 1.8.0 :-), Nothing fancy just simple dialing/trunking. > > > On Thu, Mar 10, 2011 at 11:31 AM, --[ UxBoD ]-- wrote: > >> >> -

Re: [asterisk-users] problem with crashing Asterisk 1.8

2011-03-10 Thread Peter den Hartog
1.8.0 :-), Nothing fancy just simple dialing/trunking. On Thu, Mar 10, 2011 at 11:31 AM, --[ UxBoD ]-- wrote: > > -- > > My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2. It > just keeps restarting. > Any pointers on log files to watch? I tried to

[asterisk-users] problem with crashing Asterisk 1.8

2011-03-10 Thread Peter den Hartog
My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2. It just keeps restarting. Any pointers on log files to watch? I tried to debug it but i couldn't find a good reason for the crashes. Maby the box is just overloaded or something like that but there should be a log file telling me

Re: [asterisk-users] Two Asterisk machines for redundancy

2011-02-28 Thread Peter den Hartog
Rsync to sync /etc/asterisk and use keepalived/heartbeat for a failover Asterisk IP. Make sure to read this -> http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions For "From IP rewrite" On Mon, Feb 28, 20

Re: [asterisk-users] Barge in.

2011-02-16 Thread Peter den Hartog
Feb 16, 2011 at 11:29 AM, Steve Davies wrote: > On 16 February 2011 10:13, Peter den Hartog > wrote: > > I'm running Asterisk 1.6 and was wondering if anybody have a workig > "barge > > in" solution running. > > I was thinking of using chanspy, but i wou

[asterisk-users] Barge in.

2011-02-16 Thread Peter den Hartog
I'm running Asterisk 1.6 and was wondering if anybody have a workig "barge in" solution running. I was thinking of using chanspy, but i would like that the original call would be dropped, and the new call would be the only one there. -- _

[asterisk-users] Asterisk fail over. From IP rewrite issues

2011-01-19 Thread Peter den Hartog
Hey guys, I hope somebody has some experience with the following because i'm stuck ;-). I'm creating a fail over situation for Asterisk and this works great. The only issue i have so fair os the from ip. I used the IP fix routing here -> http://www.voip-info.org/wiki/view/Asterisk+High+Availabili

Re: [asterisk-users] How to use one single IP as origination

2010-05-31 Thread Peter den Hartog
.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Groet // Kind regards, Peter den Hartog -- _ -- Bandw

Re: [asterisk-users] asterisk fax handeling

2010-03-17 Thread Peter den Hartog
5:40 AM, Peter den Hartog > wrote: > > Hello, > > I was wondering if the following was possible: > > When somebody sends a fax to my direct number 0101234567105 (my extension > > will be 105) is it possible that Asterisk, or an addon sees this as a > fax, > > and e

[asterisk-users] asterisk fax handeling

2010-03-17 Thread Peter den Hartog
Hello, I was wondering if the following was possible: When somebody sends a fax to my direct number 0101234567105 (my extension will be 105) is it possible that Asterisk, or an addon sees this as a fax, and e-mail the fax to me? So everybody with a private extension will be able to receive faxes i

Re: [asterisk-users] problems with creating a call

2010-02-10 Thread Peter den Hartog
ip of opensips and it worked.. thanks for the input tho :)! Peter On Wed, Feb 10, 2010 at 2:44 PM, Kevin P. Fleming wrote: > Peter den Hartog wrote: > > Hello, > > > > I installed Asterisk in a linonde cloud debian 5, and i'm trying to > > create a first call but wh

[asterisk-users] problems with creating a call

2010-02-10 Thread Peter den Hartog
Hello, I installed Asterisk in a linonde cloud debian 5, and i'm trying to create a first call but when i try to set up the call i see the following message: -- Called 1...@100 -- Now forwarding SIP/105-0008 to 'Local/1...@default' (thanks to SIP/100-0009) -- Executing [...@de

Re: [asterisk-users] GSM Gateway

2010-02-08 Thread Peter den Hartog
_ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digi

Re: [asterisk-users] Problems with recordings of call using Monitor

2010-02-02 Thread Peter den Hartog
wrote: > On Mon, Feb 1, 2010 at 8:55 AM, Peter den Hartog > wrote: > > I'm using the default Asterisk function Monitor to record calls, but i > have > > some issue's with this, the problem is when a call is finished, it never > mix > > in & out

[asterisk-users] Problems with recordings of call using Monitor

2010-02-01 Thread Peter den Hartog
, that's why i'm using ex...@exten. Even on a normal Asterisk machine, i have issue's with recording, i'm using Asterisk 1.6.2. Anybody got any tips on this? Thanks, Peter -- Groet // Kind regards, Peter den