On Sat, May 12, 2007 at 08:27:23PM -0500, Jadrien Wauthier wrote:
Hi,
Could anyone tell me how to read the values in the zonedata.c file?
I am looking at the zt_tone_ringtone field mainly.
{ ZT_TONE_RINGTONE, 425/1000,0/4000 }
means a tone of frequency 425 Hz for 1000 ms and then
I didn't know that !
Thanks for the tip !
2007/5/11, Noah Miller [EMAIL PROTECTED]:
Our last trial dates from 1.2.17 days (3 weeks ago).
My question is : are those HPEC audio clipping issues fixed with
1.2.17.1 ?
It's not about the Asterisk version, it's about the HPEC version.
According to
Stephen Bosch wrote:
Ariel Monaco wrote:
Dear List,
I'm having a blinking MWI light on the snom 320 even when there's no
message waiting in Asterisk.
We've managed to make the voicemail button work using
fromdomain=192.168.0.1 in sip.conf
vmexten=2500 (our VoicemailMain application extension
Atlanticnynex wrote:
whether Asterisk could handle roughly one DS3's worth of calls (672
calls) just doing the LCR (I've seen some pre-built LCR apps, looks
like they all do on-the-fly MySQL queries- I think I'd write my own
AGI that would use a cache).
When appropriately configured, MySQL
Hi,
I have been using Zapateller with a TDM400 no problems at all, but
recently I have ported our BT number to a VoIP provider, and have a
strange problem. When I phone our number I first get the BT
unavailable three tone sound, and then it actually connects the call
via IAX2.
So, I disabled
On Sun, May 13, 2007 at 01:54:08AM +0100, Chris Bagnall wrote:
3. a list of bogus entries..so when you look at it, you know it's a
fake phone number...one that recently came in that got me thinking
this was 407 111 .
I don't know much about the legal position over the other side of the
Hi
Im trying to install my TC400B trans coder card when I do:
modprobe wctc4xxp
tail -f /var/log/messages says:
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with
92 transcoders (srcs=000c, dsts=0101)
May 13 14:56:36 pbx2 kernel: Registered codec translator
Or you can specify vmexten = *97 in sip.conf and your VM button will work.
Regards,
Nitesh
Nick Adams wrote:
Stephen Bosch wrote:
Ariel Monaco wrote:
Dear List,
I'm having a blinking MWI light on the snom 320 even when there's no
message waiting in Asterisk.
We've managed to make the
Quoting Stephen Bosch [EMAIL PROTECTED]:
C F wrote:
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off.
The world is a big place, and I
I am actually getting DTMF over SIP when people call in to a clients system
that is running a2billing. They are using RFC2833.
- Original Message -
From: Remi Quezada [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
yeah that would be great! Aren't there any open-source projects out
there who handle this?
greetz
2007/5/13, Dave Bour [EMAIL PROTECTED]:
On x86 asterisk systems, there's 3 options out there, of which the
Chanskype one I've found to be the best. It's $20 US for a single
channel personal
Hi, there.
I have asterisknow beta 5 with the following data:
Ip 192.168.0.60
mask 255.255.255.0
gw 192.168.0.1
the router (a linksys) has port forwarded the port udp 5060 and from
16384 to 16482 udp-tcp from the internet to the asterisk machine.
the only protocol allowed is g729. Which work
Check rtpproxy from portone for media proxy and nat traversal.
http://www.voip-info.org/wiki/view/Portaone+rtpproxy
another option is the MediaProxy from AG projects:
http://www.voip-info.org/wiki-MediaProxy
Joss.
On 5/11/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
Hi all,
I have been using
Wow, thanks for the detailed response. For your last question, on why I
want to put a forced delay of 30 seconds, it's because the example I gave
was quite simplified to isolate what my real issue was. In reality there
are cases where I want to force 30 seconds before hanging up, and other
cases
I haven't used AEL yet, if is
ready for production at this point? (a while ago it was recommended not to
use it).
I think AEL's been replaced by AEL2 in 1.4, but I've yet to migrate our
asterisk deployments to 1.4, so cannot confirm. I've not found any stability
issue using AEL - essentially
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Yesterday we moved one of our servers to a new IP. We updated DNS and
various adapters configured to register to that server registered to the
new IP correctly. All seemed to be well.
This evening I discovered that with one exception, all of the adapters
are getting a SIP/2.0 401 Unauthorized
On Sat, 2007-05-12 at 22:35 -0600, Stephen Bosch wrote:
[EMAIL PROTECTED] wrote:
Stephen Bosch wrote:
Is Marmite also available in Ontario, or only Out West?
As far as I know, Marmite is available all across this land, from sea to
sea to sea.
Three cheers for Marmite.
IMO most
Instead of using those H323. chan drivers try using the ones in
asterisk-addons-1.2.16. They seemed to work a lot better for me than the
ones that came with the main asterisk package.
- Original Message -
From: nik600 [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
I have this in /etc/rc.d/rc.local
#Modprobe Zaptel - Loads Zaptel
modprobe zaptel
modprobe wctdm
#Start Asterisk with screen detatched
screen -A -d -m -S asterisk asterisk -vvp -c
- Original Message -
From: Josu Lazkano Lete
To: asterisk-users@lists.digium.com
Sent:
I have seen this issue where there were internet connectivity issues. Asterisk
registers every so often with the ITS. For some reason or another (it can be
many reasons such as DNS, internet, ISP has issue etc). asterisk cant
re-register so it keeps trying.
As far as the so context if you have
Are you using trixbos ? If yes have a look at the forums on www.trixbox.org
- Original Message -
From: Paul A Brown
To: asterisk-users@lists.digium.com
Sent: Friday, May 11, 2007 9:18 PM
Subject: [asterisk-users] Swissvoice IP10s setup
Hi
Does anyone have a howto on how
Thanks,
I already found these names, but maybe I missed some !
Thanks again,
JM
On 5/14/07, Yossi Ben Hagai [EMAIL PROTECTED] wrote:
Check rtpproxy from portone for media proxy and nat traversal.
http://www.voip-info.org/wiki/view/Portaone+rtpproxy
another option is the MediaProxy from AG
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