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So a friend of mine and I setup a static key based point to point
OpenVPN connection from my box to his for the express intent of carrying
IAX traffic encrypted.
His network on his lan is 172.30.1.0/24 and mine is 10.0.30.0/24. His
PBX is located at
ahdi then
> thats the probably problem
>
>
>
> On Wed, Feb 22, 2012 at 1:34 AM, Stephen Brown
> mailto:stephen.brow...@gmail.com>> wrote:
>
> On 2/21/2012 3:38 PM, isr...@gmail.com <mailto:isr...@gmail.com>
> wrote:
> > There is a bug in up to ve
On 2/21/2012 3:38 PM, isr...@gmail.com wrote:
> There is a bug in up to version 1.8.9 with external moh sources and dahdi
> timers
Do you have a link to the bug report? I was unable to find anything but
it's possible I'm not looking hard enough ;)
> Share with us your musiconhold.conf configurat
At my wits end with this, and can't proceed any further so I'm hoping
someone has seen this and can assist. I can not get streaming
musiconhold to work with Asterisk.
My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is
CentOS 5.7. When I call the musiconhold class (default fo
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Anyone know of any Asterisk 1.8 deb's available or when they might be
included in backports or (hopefully) Squeeze?
I can compile from source... but would much rather have a pre-packaged
binary if one exists...
Thanks,
Stephen
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sr --sysconfdir=/etc
> make
> make install DESTDIR=/my/destination/directory
>
> That would create the complete installation structure under
> '/my/destination/directory'
>
>
> Regards,
>
>
>
> Stephen Brown wrote:
>> I'd like to start playing wi
I'd like to start playing with 1.8, however I don't want to potentially
damage anything on my existing 1.6.2 install on my production server.
I'd like to test 1.8 against my existing configs leaving my 1.6.2
install untouched. Looking at the output of ./configure --help suggests
that it's possi
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I want to have the ability to have anonymous SIP calls hit my server but
I want to run it on different ports and create an SRV record for my
target domain.
My understanding of SIP is limited, but in a nutshell I want to
accomplish the following:
- -
Ditto I'm running a Supermicro Atom based dual core server and it's rock
solid!!!
These make excellent servers for Asterisk installation IMHO.
On Fri, Jun 11, 2010 at 05:42, --[ UxBoD ]-- wrote:
>
> - Original Message -
> > On Thu, 10 Jun 2010, Michelle Dupuis wrote:
> >
> > >
> You only get past this point when this call is over.
That makes sense now that I think about it.
> Can the CO send Flash to the CPE?
I am the CO in this case :)
I started looking at the "D" option of the dial command, is there any
way to send a DTMF digit followed by a hookflash at call origina
I've got a piece of CPE equipment that has an FXS port that I have tied
to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a
standard telephone connected to it, I get a dialtone. If I dial a digit,
and send a hookflash, the device will provide a dialtone back for the
next av
t;) in new stack
[Sep 25 13:34:56] -- Executing [...@macro-get-vmcontext:1]
Set("DAHDI/1-1", "VMCONTEXT=default") in new stack
[Sep 25 13:34:56] -- Executing [...@macro-get-vmcontext:2]
GotoIf("DAHDI/1-1", "0?200:300") in new stack
[Sep 25 13:34:56]
risk-users-boun...@lists.digium.com] *On Behalf Of
*Stephen Brown
*Sent:* Friday, September 25, 2009 10:41 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] DAHDI disconnect supervision timing
I have, but I wanted to see if I could fix this problem before I
started experimenting with
ssages
> if that gets ever detected
>
> also you may try hanguponpolarityswitch=yes in chan_dahdi.conf
>
> Martin
>
>
> On Fri, Sep 25, 2009 at 10:19 AM, Stephen Brown Jr
> wrote:
>
>> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
&g
:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Stephen Brown Jr
*Sent:* Friday, September 25, 2009 10:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] DAHDI disconnect supervision timing
Ok
Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS line.
I have asked for "disconnect supervision" to be provisioned o
Following up on this with some testing I have done this afternoon. I
have lengthened the default ringtime to 25 seconds from 15, and have
tried both kewlstart and loopstart. Kewlstart appears to be the most
functional, if I hang up before it hits app_voicemail, it appears to
work as intended, a
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Stephen Brown Jr
*Sent:* Tuesday, September 15, 2009 11:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] DAHDI hangup detection
lco may or may not be of assistance, but a simpler workaround might be to
> hangup on silence or query the line through AMI before transferring to
> voicemail.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.
I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled
from source on a Debian Lenny box, also running FreePBX 2.5.2. I also
have an OpenVOX TDM400 card installed with an FXO port on port 1, and an
FXS port on port 2. I have a POTS line installed and working on the FXO
port.
Howe
you could check this with the etchandahalf (2.6.24) kernel?
That was going to be my next step if I couldn't get it resolved
Thanks,
Stephen
On Thu, Dec 18, 2008 at 11:57, Tzafrir Cohen wrote:
> On Thu, Dec 18, 2008 at 10:49:03AM -0500, Stephen Brown Jr wrote:
>> I'm having troub
I'm having trouble with ztdummy and I can't seem to figure it out. I
am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates
applied and I have compiled Zaptel from source along with a new kernel
from Debian sources to include 1khz timer support.
The modules build fine, yet when I load the
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