Links to my configuration files for the polycom phone. As you'll see, majority
of my settings are default. Hope it will help you to determine where my
problem is at.
MAC Address cfg file
http://docs.google.com/Doc?id=dggrkn86_2dc3qfdgr&hl=en
Extension cfg file
http://docs.google.com/Doc?id=d
I just tried it from a stick, nice looking but too many bugs:
camera and mic don't work: so useless for conversation.
https://answers.launchpad.net/ubuntu-eee/+question/43167
and that was posted 2008-08-26 now is Nov. 15/08 and it still doesn't work, so
no mic no conversation :-/
Some suggested
Alex Balashov wrote:
> The solution for the problem of an IAX client is a SIP client.
>
That's not a particularly good solution if you have a NAT between your
client and Asterisk. IAX is still *much* easier to get working through
a firewall.
___
-
At 21:06 11/15/2008, hin lee wrote:
>Here are more information as requested:
>
>Asterisk v. 1.4 (running PBX in a Flash)
>Using Zaptel, TDM800P card
>Polycom running: 3.03 SIP Firmware
>Provisioning by: FTP
>
>I am calling from my Polycom to other land line phones. Hope I
>provided enoug
Here are more information as requested:
Asterisk v. 1.4 (running PBX in a Flash)
Using Zaptel, TDM800P card
Polycom running: 3.03 SIP Firmware
Provisioning by: FTP
I am calling from my Polycom to other land line phones. Hope I provided enough
information.
Thanks!
Hin
--- On Sat, 11/15/08, D
I installed eeeUbuntu ( http://www.ubuntu-eee.com/ ). I followed
instructions on their wiki to create the USB-stick installer from another
Ubuntu PC that I have. It looks like they've made the process even easier
now, with a GUI-based application that will prep a USB stick with any ISO
that you cho
The solution for the problem of an IAX client is a SIP client.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
___
-- Bandwidth and Colocation
On Sun, Nov 16, 2008 at 9:37 AM, Tony Mountifield
<[EMAIL PROTECTED]>wrote:
>
>
Actually, if Read() works, then WaitExten should have worked too. I expect
> what was missing was the Answer(). So this ought to work as an alternative:
It doesn't for some reason... I end up getting the timeout call
Well, I've tried to find Ubuntu but so fare I'm not sure which one. I have eee
pc 1000 (one with 40GB SSD so plenty of room for any modern distro.
Which ubuntu did you loaded? It has to be something that loads onto USB
bootable stick (and not through Windows as I don't have one).
--
#Joseph
GPG
Hi Joseph,
Not directly related to your question (it's more an answer for the
"something better" part of your plan), but I've loaded Ubuntu onto my Asus
eeePC 4G Surf, and I've found that ZoIPer works pretty well.
Cheers,
AR
--
Alex Robar
[EMAIL PROTECTED]
On Sat, Nov 15, 2008 at 5:49 PM, Jos
Joseph wrote:
>
> It keeps complaining about /lib/tls/libc.so.6 'GLIBC_2.4' not found.
>
> How do you install this library on EEE pc Xandros? (I know Xandros is Debian
> based) but this is eee pc.
>
>
You should ask on another list but this should get you started;
http://forum.eeeuser.com/viewt
On 11/15/08 18:04, David wrote:
>Joseph wrote:
>> What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
>> software)?
>>
>> I'll eventually replace this crippled Linux with something better but I
>> don't time to play around with it as most divers and modules are still too
>> new
This would be better on the Asterisk-gui list, but it's because it's written in
users.conf
registersip=yes
-bk
- Original Message -
From: "Joseph L. Casale" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, November 15, 2008 4:18:25 PM
Joseph wrote:
> What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
> software)?
>
> I'll eventually replace this crippled Linux with something better but I don't
> time to play around with it as most divers and modules are still too new and
> not fully available in all distros
Hi,
I'm noticing MixMonitor records 5 seconds aprox less of a call.
The recording is iniciated via Queue and ends at the hungup.
(gsm format), when I listen to the audio file, has 5 seconds missing at the
end of the call.
Any idea??
thanks
ASt.1.6.0.1
__
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)?
I'll eventually replace this crippled Linux with something better but I don't
time to play around with it as most divers and modules are still too new and
not fully available in all distros.
--
#Joseph
GPG KeyID:
In article <[EMAIL PROTECTED]>,
Mikel Lindsaar <[EMAIL PROTECTED]> wrote:
>
> Good. Got the darn thing working.
Good news! Been away from the computer for a while, so wasn't able to
keep up with your efforts.
> Problem was the NEC Xen Master does post-connect DTMF to dial. So I had to
> read t
I was playing with 1.6.0.1 and the latest gui and wondered how my sip
did was registered after creating it? How does this take place, normally
I made a register => command in sip.conf but don't see this in any files?
Thanks!
jlc
___
-- Bandwidth and Col
Actually, it could be within Asterisk, but only if you have Zaptel
hardware. If you are only using SIP devices, then the problem is with
the phone configuration. You really don't provide enough information to
determine what is causing your problem. How are you provisioning the
phones? What
Good. Got the darn thing working.
Problem was the NEC Xen Master does post-connect DTMF to dial. So I had to
read the digits after connect.
Then, I had to configure the PRI to be pridialplan = unknown
Thanks for your help Tony.
Here is the end result for Google's sake
#
Hi,
I'm noticing MixMonitor records 5 seconds aprox less of a call.
The recording is iniciated via Queue and ends at the hungup.
(gsm format), when I listen to the audio file, has 5 seconds missing at the
end of the call.
Any idea??
thanks
ASt.1.6.0.1
__
Probably has nothing to do with Asterisk. You can set the volume and
persistence in the phones config files.
Michael
On Fri, 14 Nov 2008 22:43:45 -0800 (PST), hin lee wrote:
>Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't
>figure out why the volume is so low. Ho
OK, made some progress.
With some help from friends on #Asterisk, found that the NEC was doing post
connect dialing.
So then, added the following to extensions.conf:
[from-nec]
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(digit)=2)
exten => s,n,Set(TIMEOUT(response)=5)
exten => s,n,Read(DialedN
On Sun, Nov 16, 2008 at 3:13 AM, Mikel Lindsaar <[EMAIL PROTECTED]> wrote:
> On Sat, Nov 15, 2008 at 11:05 PM, Tony Mountifield <
> [EMAIL PROTECTED]> wrote:
>
>> In article <[EMAIL PROTECTED]>,
>> Mikel Lindsaar <[EMAIL PROTECTED]> wrote:
>> > I have an NEC PBX connected via a TE210p E1 line to a
On Sat, Nov 15, 2008 at 11:05 PM, Tony Mountifield <[EMAIL PROTECTED]
> wrote:
> In article <[EMAIL PROTECTED]>,
> Mikel Lindsaar <[EMAIL PROTECTED]> wrote:
> > I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box.
> > NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco
> > In
In article <[EMAIL PROTECTED]>,
Mikel Lindsaar <[EMAIL PROTECTED]> wrote:
>
> I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box.
>
> NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco
>
> Incomming calls from the telco to the asterisk box to the NEC work fine with
> indi
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