Is there a trade-in program in place ? I have a te410p and a te405p that
I am not using because of various problems we had, but would like to try
the te407 ...
Julian
BJ Weschke wrote:
On 11/30/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Thursday 23 November 2006 11:44, Heidi Mendoza
Hi
On Sat, Dec 02, 2006 at 07:17:22PM -0500, Matthew Rubenstein wrote:
On Sat, 2006-12-02 at 09:53 -0700,
[EMAIL PROTECTED] wrote:
Date: Sat, 2 Dec 2006 11:51:37 +0200
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] zaptel compilation problems with linux
On 2 Dec 2006, at 16:59, Alex Rixhardson wrote:
Hi guys,
Here is a bit more detailed information of my problem:
If I connect Asterisk PBX to the Polish telco via E1, I don't get
any red alarms or anything. The line seems to be fine and the
inbound calls are also accepted by the Asterisk.
Hi List,
I just wanted to let eveyone that RNK uses asterisk. One more big company to
let clients know that asterisk is great.___
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Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power??
Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power??
Dec 2 17:45:19 WARNING[64722]: translate.c:133 ast_translator_build_path: No
translator
path from gsm to unknown
Dec 2 17:45:19
I know this has been asked before and I went over the wiki but I have not been
able to come to a clear answer.
1) If I have SIP Provider Asterisk - ATA and vice versa (ATA -
Asterisk SIP Provider) from what I understand if NO NAT is being used
then asterisk just starts and
If I have SIP Provider (sending call as G729) Asterisk (picks up and asks
for PIN) --- call sent to SIP provider I understand that I will need a
g729 liscence for when the user needs to enter thier PIN number. Once the call
is handed off to the SIP provider will I still need a liscence
Asterisk wont sit in media path if both callee and caller agrees on common
codec, both have canreinvite=yes in sip.conf, no t,T are used in dialplan (
please correct me if i am wrong ) , no call recording is enabled .
I think asterisk does native bridging even if one is behind nat ( i tested
This is correct, if no NAT is involved anywhere and reinvites are
allowed then Asterisk will stay out of the media path and be used only
as Signaling server. So as for your answer yes, it will be able to
handle more calls than expected because there is no CPU overhead of the
media path.
It
I wonder if anyone has experienced an issue I have found with the
Linksys SPA-841 phone.
On my Asterisk (Trixbox 2), to login to a queue, a user must enter the
queue number, followed by the * key. This works fine on my Companies
mix of phones, with the exception of the Linksys (Sipura) SPA-841.
Hi All,
Has anyone else noticed that when a sip phone sitting behind a nat
registers to asterisk using realtime database, the private IP of the
phone is put into the fullcontact field instead of the public contact
IP. The database has the correct public IP in the ipaddr field and
correct port
Not very good at answering followups to your ads, are you, Sam?
On 01/12/06, Peter Bowyer [EMAIL PROTECTED] wrote:
On 30/11/06, Sam Tam [EMAIL PROTECTED] wrote:
We do have @cough VoIP GSM Gateway for sell as well @ cough
Try to search on ebay for gsm voip gateway and you will see some in
Have you looked at his website, www.netenable.co.uk ? Looks like he pays bills
the same way as he answers followups ;-)
g
-Original Message-
From: [EMAIL PROTECTED] on behalf of Peter Bowyer
Sent: Sun 03-Dec-06 8:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi List,
I am experiencing an issue with a server running asterisk; I installed an
AVM FRITZ card and configured it to work with the capi module.
Once everything is installed the card works perfect; the issue is that every
time I reboot the machine I have to re install the capi4k-utils
Hi All,
To coincide with my Asterisk 1.4 Beta Howto, I've also started the
spandsp3 howto. The howto is functioning, however when attempting to
use rxfax or txfax it crashes asterisk. I am not a programmer thus
couldn't figure out how to make it not seg. Check out the WIP howto
for more
A while ago I wrote a numbers guessing game to keep me entertained on
those really boring days :). I've uploaded it to the blog for the rest
of you to enjoy or modify as well. It's a simple game to guess what
number the PBX is thinking of and return a yay you got it right or a
sorry that's wrong.
Thanks for your help Claudemir, I look forward to the response. Seems
odd that they don't post an archive of their old firmware versions on
their website, or at least ones that are required to get to the latest
release from whatever is in the field already.
Regards,
Scott
Hi everybody,
I want to know how to get the uniqueid or a call started from asterisk
manager using Originate command.
Best regards
Rodrigo Gonzalez
___
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To
I have a SIP carrier which accepts only G729 connections from my
Asterisk server. If all the server does is Dial() (out) two legs of a
call which are natively bridged, with no processing the media (and no
DTMF detection, etc), do I need to install a G729 codec of my own? All
the media from
Thanks. That looks exactly what I need. Will test in the next couple of
days
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Saturday, December 02, 2006 8:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Not sure if anyone has posted this before, but it would be great
(providing it works well) to connect this to an FXO port.
http://www.jr.com/JRProductPage.process?Product_Code=AEE+USB07051C01JRSource=PriceGrabber.datafeed.AEE+USB07051C01
Anyone tried this product yet?
Great price for a cool
Hi, iam new in this milis
I have problem with TDM01B Installation,
output zttool command is
Unable to open /dev/zap/ctl: No such device or address
and then i find the same IRQ uses VGA compatible controller and
Communication controller is 169
What can i do next ?
please your advice
I think you do need to buy the G729 for each call. If your system is
using anything other than G729.
That is the way I was told it works. But I don't use G729.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VOIP PBX) 1-866-638-1254
(Voip PBX) Free World
I think you do need to buy the G729 for each call. If your system is using
anything other than G729.
True, you only need a CODEC if your system is using
anything other than G729. In that case you would
only be using pass through.
Only if you are COding or DECoding do you need a CODEC.
For
In reality, this is the one I've found that has exactly what our client
needs. However, it seems to be a closed system so we are evaluating it
further.
AstBill and MOR don't seem to have the feature to offer referral credits
out-of-the-box. Maybe we missed something?
Thanks,
Daniel
I am wondering if there is any such thermostat available which can be
controlled from Asterisk. Like you call your home pbx, dial some extension,
e.g. 333 and it asks to set the temperature, you enter a temperature, and it
sets the thermostat to that temperature. This thermostat will be very
Hello Users,
Nice to meet You,
How can i Processed the call Snooping, it my fifth Requesting and posting
to Users, Nobody replies it,,,
in Call snooping , How can i record the Voip users,
We can record to users by using Monitor Application ,
Can any give clear layout of the Snooping
On Mon, Dec 04, 2006 at 09:12:21AM +0700, Mochamad Susantok wrote:
Hi, iam new in this milis
I have problem with TDM01B Installation,
output zttool command is
Unable to open /dev/zap/ctl: No such device or address
and then i find the same IRQ uses VGA compatible controller and
hi all,
i need to integrate and modify one of the application in asterisk/apps
section...
whenever i modified small steps ..in order to check and compile i 've to do
recompile the whole asterisk module and it consuke to much time...
please anyone couls you tell me, how can i modify it ,
On Monday, November 27, 2006 10:23 AM Louis-David Mitterrand wrote:
Hello,
With the following setup:
- asterisk 1.2.13,
- zaptel 1.2.10
- bristuff 0.3.0-PRE-1v
- quadbri card,
Have you tried using bristuff 1v with the qozap driver of 1s? All qozap
versions after 1s had serious
Hello ppl,
Am using realtime odbc storage for voicemail, sip users/peers, static
for extensions and so on.
My issue is I am not getting MWI for any fones, even tho I've got
rtcachefriends=yes in sip.conf
WIth tcpdump, I always see the NOTIFY going as
Messages-Waiting:.no
Never tried, but this should work: http://www.smarthome.com/3001.html
Lots of neat stuff on that site.
On 12/3/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I am wondering if there is any such thermostat available which can be
controlled from Asterisk. Like you call your home pbx, dial some
On Mon, Dec 04, 2006 at 09:12:21AM +0700, Mochamad Susantok wrote:
Hi, iam new in this milis
I have problem with TDM01B Installation,
output zttool command is
Unable to open /dev/zap/ctl: No such device or address
and then i find the same IRQ uses VGA compatible controller and
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