Re: [asterisk-users] Site to site VPN problems

2019-12-03 Thread Jan Bakuwel
Hi Ira, On 4/12/19 5:12 pm, Ira wrote: > Re: [asterisk-users] Site to site VPN problems Hello Jan, > > Tuesday, December 3, 2019, 2:34:46 PM, you wrote: > > > /Jan> localnet=192.168.1.0/255.255.255.0 > Jan> localnet=192.168.3.0/255.255.255.0 > > Jan> Just making sure you have both local subnets

Re: [asterisk-users] Site to site VPN problems

2019-12-03 Thread Jan Bakuwel
Hi Ira, On 4/12/19 10:59 am, Ira wrote: > Re: [asterisk-users] Site to site VPN problems Hello Jan, > > Tuesday, December 3, 2019, 1:39:43 PM, you wrote: > > /Jan> Did you include both 192.168.1.0 and 192.168.3.0 in the localnet > Jan> declarations in sip.conf? > > /I did, both look like this: >

Re: [asterisk-users] Site to site VPN problems

2019-12-03 Thread Jan Bakuwel
Hi Ira, On 4/12/19 10:31 am, Ira wrote: > Site to site VPN problems Hello Asterisk, > > Sorry if this is off topic. > > I have recently acquired a second site and I'd like to put a couple of > phones registered to me Asterisk server there. There is a Site to Site > OpenVPN  192.168.1.0 connecting

[asterisk-users] ExtenSpy - no audio

2013-01-31 Thread Jan Bakuwel
Hi, Does anyone have any experience debugging the ExtenSpy function? Asterisk 1.6 (yes, I know it's old) on Debian. core show channels: Channel Location State Application(Data) SIP/570-00031ac1 808@monitor:1Up ExtenSpy(808@desks,w)

Re: [asterisk-users] [SOT] Virtualising Asterisk

2011-05-09 Thread Jan Bakuwel
Hi Phil, Happily running with the following here: dom0: Debian Lenny Xen 3.2-1 2.6.26-2-xen-amd64 domU: Asterisk 1.4 Debian Lenny 2.6.26-2-xen-amd64 domU: Asterisk 1.6 Debian Squeeze 2.6.32-5-amd64 (which is a Xen-aware kernel) domU: Asterisk 1.8 Debian Squeeze 2.6.32-5-amd64 (which is a

Re: [asterisk-users] asterisk practices

2011-04-28 Thread Jan Bakuwel
Hi Vip, On 28/04/11 05:34, vip killa wrote: I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call -

Re: [asterisk-users] Asterisk 1.6.2.18 Now Available

2011-04-28 Thread Jan Bakuwel
Hi, I'm about to deliver a production system based on Debian Squeeze and Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8 packages for Debian Ubuntu are available from packages.asterisk.org. Observing some recent discussions on this list, it seems that 1.8 might not yet be

Re: [asterisk-users] Asterisk 1.6.2.18 Now Available

2011-04-28 Thread Jan Bakuwel
Hi Matt, On 29/04/11 11:26, Matt Riddell wrote: On 29/04/11 11:19 AM, Jan Bakuwel wrote: Hi, I'm about to deliver a production system based on Debian Squeeze and Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8 packages for Debian Ubuntu are available from

[asterisk-users] Retaining original caller id

2011-04-26 Thread Jan Bakuwel
Hi, I'm trying to get my head around an interesting problem (well I think it's interesting :-) ). An inbound call (say from extension 100) gets send to a queue and one of the members (say on extension 200) answers the call when her/his phone rings. In this case ${CALLERID(num)} = 100 and