We Use both Grandstream and Aastra phones
Some simple scripts improve the Grandstream configuration ease of use
but the Aastras use a straight text file and are much better documented.
regards,
Drew
Ahsan Masood wrote:
Hi,
We are using following phones for large deployments using auto-prov
On 2/8/07, Remzi Semsettin Turer <[EMAIL PROTECTED]> wrote:
This is a solution if your provider is using IAX, but we are stuck with
SIP.
Huh? What do the two have to do with each other?
___
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as
Chris, (or others), do you have any negative experience with Thomson
2030? it looks very promising!
I hesitate between thomson and linksys spa 922/942,
I'm not sure, what is better for bussines use :-\
snoms are probably also good, but functionality/price ratio is, imho,
better for thomson or li
our Polycoms reregister almost immediately. I think the problem your
running into is that when the softphone is registered the polycom gets
some kind of error from asterisk which prevents it from reregistering
Rob Schall wrote:
That's what I would have thought. I set the timeout to be 300 secs
This worked. Great and thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Wednesday, February 07, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers
On Thursday 08 February 2007 07:32, [EMAIL PROTECTED]
wrote:
> Does anyone have any recommendations for a phone that has easy to
> understand/implement central provisioning? I've used CISCO 79XX phones,
> and they're great (but too expensive). I like Grandstream phones, but
> their provisioning su
This is a solution if your provider is using IAX, but we are stuck with SIP.
I find it surprising that txfax and rxfax not compiling under 1.4, but oh well.
Warm Regards,
Remzi Turer
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ardjan Zwartjes
Sent:
> Asterisk is getting red alarms on my T1, sometimes once or twice a
> day, but today it happened 5 times. Even once is too many. Every
> call in progress is dropped.
Red alarm means that the hardware is not seeing the T1 signal coming in.
This most likely is a cable or wiring or perhaps a har
We have a similar situation and we do a realtime lookup in an external
db, works like a champ
Steve Murphy wrote:
On Wed, 2007-02-07 at 22:21 -0500, Lee Jenkins wrote:
Eric Germann wrote:
We're beginning to test MultiTech's CallFinder CDMA Units, one for Sprint
PCS and one for Alltel.
On Thu, 2007-02-08 at 10:32 -0200, Paulo Vicentini wrote:
> Hi
>
> I set up call back functionally thru AMI (local channel).
>
> The two calls are bridged and the call is established.
>
> But when I hang up the local channel (the first extension that rang),
> the other leg of the call *is not
I'm tempted to rebuild my asterisk network with AsteriskNow - my
question is, can you ADD anything to it? i.e. cdr_mysql logging? I thought
I saw it didn't have that
And how does it handle the hardware? I don't use digium cards in all of my
servers because of country issues (Junghanns in Ge
It's not that Digium don't want fax or t.38 support, it's just that it is
not very likely for Steve Underwood to provide it for Asterisk. I'm sure
that Digium are very keen for someone to write and contribute t.38 code for
Asterisk, it's just that there aren't very many people with the required
Is it possible to use Digium (or Sagnoma, or Beronet) cards with Asterisk on
Vmware?
Has anyone done it?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
__
I must clarify my original message. Maybe
confusion is due to my poor english. So I'll make a list of statements:
- Each ISDN line in Italy can be splitted in two analog lines
- You can use those analog lines as normal analog lines
- I have already invested in analog hardware (my
fault of cours
That's what I would have thought. I set the timeout to be 300 secs, but
the phone never seems to re-register. We could do a group dial, but like
you said, there would be a lot of errors in the log, which we are trying
to avoid. Has anyone been able to get a polycom 501 to re-register
itself without
Yuan Liu wrote:
My multiple postings to this list this morning got garbled in
http://lists.digium.com/pipermail/asterisk-users/, and don't come back from
list. (e.g.,
http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I
thought it was Hotmail, so I saved one outgoing
Which H.323 channel driver are you using, and could you post a log or debug
of a session.
Craig
- Original Message -
From: "Andrei U" <[EMAIL PROTECTED]>
To:
Sent: Thursday, February 08, 2007 2:41 AM
Subject: [asterisk-users] H323 to SIP - One way voice
Hello all,
I want to use a
On Thu, 2007-02-08 at 13:55 +0100, Tomislav Parčina wrote:
> In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> > Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
> > before t.38 is ever utilised, not even pass-thru.
> >
> > 1.4 Adds support for T.38 pass through only an
Set a variable that you can then use GotoIf in the dialplan to branch to the
required exten
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: prasanth <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Thursday, 8 February, 2007 10:06:07 AM
Subject: [asterisk-user
Lookup 'articulation'
On Feb 1, 2007, at 1:53 AM, Dovid B wrote:
Anyone know of a softphone for the Palm OS ?
Thanks.
Dovid
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asterisk-users mailing list
To UNSUBSCRIBE or update options visi
Hi,
I'm new to *,so i apologize for stupid questions.
I'm having problem with this arhitecture:
I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm
using GSM codec.
In extensions.conf I have:
exten => 337,1,Dial(SIP/99@)
so when i dial 337 from sjphone Asterisk is colling 9
i started asterisk my typing the command #/usr/sbin/asterisk -c but it is giving error that it couldn't establish connectiom with mysql. failed to connect database server superswitch on 192.168.1.205 unable to get our IP address , Skinny disabled. please help
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
> before t.38 is ever utilised, not even pass-thru.
>
> 1.4 Adds support for T.38 pass through only and no other sort of
> faxing, the endpoint must support T.38 and you m
Am 08.02.2007 um 13:02 schrieb Benito Camelas:
To someone who have done the dCAP exam.
I did the dCAP a couple of weeks ago.
I would like to know about it: test and practises questions examples,
difficulty level,... I'll be very grateful if somebody sends me an
exam model.
The practical tes
On Tue, Feb 06, 2007 at 09:41:30AM +, Tim Panton wrote:
>
> On 5 Feb 2007, at 21:46, chester c young wrote:
>
> >Need to deploy between 50 to 300 lightweight Linux - only browser
> >and softphone.
>
> You might want to consider our lightweight java softphone (Corraleta
> SDK) - it can be
Hi
I set up call back functionally thru AMI (local channel).
The two calls are bridged and the call is established.
But when I hang up the local channel (the first extension that rang), the
other leg of the call *is not released*
Time events:
0) Socket communication(AMI)
1)extensionA r
Hello.
To someone who have done the dCAP exam.
I would like to know about it: test and practises questions examples,
difficulty level,... I'll be very grateful if somebody sends me an
exam model.
Thanks in advance
___
--Bandwidth and Colocation provide
Many thanks Stefan!
It works like a charm...
Kind regards,
Pierre
===
Write a cronjob which creates a call file. Shouldn't be a big thing.
In case you are not familiar with call files: Create a file dummy.call
with the following content.
---cut---
Chan
Hi,
We are using following phones for large deployments using auto-provisioning.
Grandstream phones (full range)
Snom Phones (full range)
Aastra Phone (full range)
UTstarcom (Wifi phones)
~Ahsan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christop
The part about 4569 being the IAX2 setup port, is not correct.
All traffic, including RTP, travel through this port, when you use IAX.
rtp.conf is used for SIP traffic, and possibly H232.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed W
Sent: 8. f
Hi there,
As described on voip-info here
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue, if I use
realtime queues, alterations to the list of members don't alter until a
new call joins the queue.
Is there anything I can do about this? I've tried looking for a bug
number, but to no
Hi all,
I'm trying to send FAX with an anolog fax behind a Patton M-ATA to an
other analog fax plug on directly on the PSTN network.
I use the last stable version of Asterisk 1.4 ...
Somebody have any information why it's doesn't work a all ?
Thanks a lot,
Thomas
_
Hi
Yes, I know that I am using IAX2 and not SIP for my connection to
teliax. IAX2 is the preferred protocol for connection to teliax. I
have the firewall configured to prioritorize port 4569 for IAX2.
1) 4569 is only the IAX setup port. Edit rtp.conf to limit the rtp
ports to some subset
Check from the sites in question using "testmyvoip.com" or whatever the
site is called.
In the UK I found that some strange things sometimes happen. At one
point I was sure that BT were perhaps misclassifying IAX packets as
P2P... However, not had a problem with SIP.
Beware that ADSL uses v
Digium support cleared the issue for me, they sent me a new "register"
utiliy by mail and this one worked as expected. I registered my codedc
and tested my codec. If anyone needs to know, I tested the codec using a
SIPURA 3000 ATA so I can confirm this ATA works with G729.
I'd like to add: Dig
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm using a Thomason ST2030. Had difficulties in the beginning, but
after a firmware upgrade it works fine. And autoprovisioning works good.
Most of the parameters are described in their official (marked as
confidential) admin documentation from t
Paul Hales wrote:
I know Brett and Jurgen have been pretty happy with the Snom's - Brett
even wrote an auto-provision utility for the Snom's at one time.
Yes, look at the latest Trixbox for the basic SNOM templates and then
off you go.
You setup a tftp server (easy), the phone looks for t
> "CB" == Chris Bagnall <[EMAIL PROTECTED]> writes:
CB> I have run a few speed tests from the sites in question (iperf to
CB> the machine in the datacentre) and I'm consistently getting around
CB> 380k upstream and 5.5mbit downstream, even during peak hours. Some
CB> distance away from the quo
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Hi people.
I'm hoping someone has come across this problem with version 1.2.14
In my dial plan I call various SIP phones using the following little
macro:
I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I
execute AGI in java which plays few wav files depending on external
parameters.
Can I have a dial plan inside my AGI? If not, how do I accomodate user
who needs to reach extension 2 from my agi? I have tried stream file and
g
> Need to deploy between 50 to 300 lightweight Linux - only browser
> and softphone.
You might want to consider our lightweight java softphone (Corraleta
SDK) - it can be embedded in
a web page - zero install/config in the client. The UI is in HTML and
javascript,
so you can get it _exactly_ the
I know Brett and Jurgen have been pretty happy with the Snom's - Brett
even wrote an auto-provision utility for the Snom's at one time.
later,
PaulH
On Thu, 2007-02-08 at 19:45 +1100, Rod Bacon wrote:
> Does anyone have any recommendations for a phone that has easy to
> understand/implement cen
Hi all,
I am looking for a solution for the following problem.
I have a little callcenter with 20 agents and 20 incomming analog lines, one
for each agent. I need to have abailable as incomming analog lines (FXO
Ports) as agents logged, not all the agents are logged all the time. It is
needed for
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> Hi,
>
> I just want out find out how to do bill recon's when you send calls to a
> provider. They send me
> their CDR's, and when I compare it to my * CDR's, some calls are 1 second
> off, either way.
> How in general is it done by ot
On Wed, 7 Feb 2007, Mark Coccimiglio wrote:
Ok here is a real geek question,
I building my own linux kernel for my asterisk system and came across the
kernel setting for the timer frequency. I have one of 3 hardcode choices
100Hz, 250 Hz and 1000Hz. From what I understand the default Freq w
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> What do you mean by mapping the 200 ?
>
> In this example I can pickup any ringing extension:
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
>
> "If phone with number 42 rings you can catch the call by dialing 742. You
> don't
Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I've used CISCO 79XX phones,
and they're great (but too expensive). I like Grandstream phones, but
their provisioning sucks.
What is everybody else using in large environments where ind
I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the
topology:
VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
|
FXO ___ PSTN extension
When I call a VoIP extension on that box
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