Displaying 18 results from an estimated 18 matches for "innovax".
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2004 Sep 16
2
Help with E1 configuration
Hi,
I currently have a E100P card installed on my machine and the E1
subscription will be activated pretty soon. However, I have no idea how to
configure asterisk to make inbound and outbound call using the E1.
Especially for extensions.conf. Below is the configuration I used for
zaptel.conf and zapata.conf. Is it possible if someone can verify if the
configuration for zaptel and zapata is
2005 Mar 08
3
NAT Far End Traversal
Hi List,
After some research, it seems the only reasonable thing to do in order
to get SIP phones behind NAT working reasonably well without fiddling
with the DSL router is to have some kind of far end nat traversal mechanism.
Is there any way to do this with open source tools? I've seen somewhere
that far end nat traversal can be achieved with SER + nathelper does the
job... has anybody
2003 Aug 19
0
Re: Open source IP phone, maybe?
I concur with Jose. The Atmel AVR series packs a lot of bang for the buck. They
also come in a 3.3v low power version for use in battery powered systems.
Gene
-----Original Message-----
From: Leo Ann Boon [mailto:leo@innovax.com.sg]
Sent: Tuesday, August 19, 2003 7:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: Open source IP phone, maybe?
Ubicom's Scenix IP2K. Sxdesign has an SIP phone platform using that chip
http://www.sxdesign.com/index.php?page=solutions&submnu=voip
Jose Ild...
2003 Apr 15
0
Re: Asterisk-Users digest, Vol 1 #286 - 14 msgs
...on non-intel hardware (Adam Goryachev)
> 11. Re: Sip & Intercom (James O. Sizemore III)
> 12. * kernel panic (Tamas Levente)
> 13. Re: * kernel panic (Steven Critchfield)
>
>--__--__--
>
>Message: 1
>Date: Mon, 14 Apr 2003 17:06:56 +0800
>From: Leo Ann Boon <leo@innovax.com.sg>
>To: asterisk-users@lists.digium.com
>Subject: Re: [Asterisk-Users] Conferences without zaptel devices
>Reply-To: asterisk-users@lists.digium.com
>
>Just modprobe zaptel to load the ztdummy module. I believe you'll need a
>recent CVS snapshot of zaptel for this to...
2003 Apr 15
0
Re: Asterisk-Users digest, Vol 1 #286 - 14 msgs
...on non-intel hardware (Adam Goryachev)
> 11. Re: Sip & Intercom (James O. Sizemore III)
> 12. * kernel panic (Tamas Levente)
> 13. Re: * kernel panic (Steven Critchfield)
>
>--__--__--
>
>Message: 1
>Date: Mon, 14 Apr 2003 17:06:56 +0800
>From: Leo Ann Boon <leo@innovax.com.sg>
>To: asterisk-users@lists.digium.com
>Subject: Re: [Asterisk-Users] Conferences without zaptel devices
>Reply-To: asterisk-users@lists.digium.com
>
>Just modprobe zaptel to load the ztdummy module. I believe you'll need a
>recent CVS snapshot of zaptel for this to...
2003 Oct 14
5
Digium cards just for timing
Hi,
I've found that neither Michael Manousos patch nor ztdummy driver
do not fix musiconhold sound interruption problem up to acceptable quality
level. Sound is choppy here anyway.
It is my understanding (please correct me if I'm wrong) that if I have
a Digium card in my asterisk machine, these problems should be gone
'cause those cards provide some reliable timing. So I have no
2004 May 20
4
x100p card + dailing out
I think I have it configured properly. ztcfg -vv shows it as channel 1 and
zttool shows it as OK. But I can't dial out.
When I try, it shows it arrive in teh right stack, but then issues the
following errors:
channel.c:1676 ast_request: No channel type registered for '{PSTN-1}'
app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}'
= = Everyone is busy at
2003 Apr 15
9
Extensions.conf
...on non-intel hardware (Adam Goryachev)
> 11. Re: Sip & Intercom (James O. Sizemore III)
> 12. * kernel panic (Tamas Levente)
> 13. Re: * kernel panic (Steven Critchfield)
>
>--__--__--
>
>Message: 1
>Date: Mon, 14 Apr 2003 17:06:56 +0800
>From: Leo Ann Boon <leo@innovax.com.sg>
>To: asterisk-users@lists.digium.com
>Subject: Re: [Asterisk-Users] Conferences without zaptel devices
>Reply-To: asterisk-users@lists.digium.com
>
>Just modprobe zaptel to load the ztdummy module. I believe you'll need a
>recent CVS snapshot of zaptel for this to...
2004 Mar 16
24
Softfax/spandsp
Hi all,
After a long time having no time, I have finally done some fresh work on
my software fax machine. I have replaced the original carrier tracking
with something more robust. I have also added 4800, and 2400 bits per
second modes, and cleaned up a few bugs in areas like superfine mode
operation. I apologise for this update taking so long.
At ftp://ftp.opencall.org/pub/spandsp you will
2004 Mar 31
2
ANNOUCEMENT: Tool to combine call recording into a single 'stereo' file
Figured this might be interesting to some on the list. I've written a
small tool to combine the 2 res_monitor created wav files into a single
compressed (IMA ADPCM 8KB/s) stereo file. During playback, you can
select the party to listen to by fiddling with the audio balance.
Email me off-list if you would like a copy.
Cheers.
-Leo
2004 Apr 04
1
SetCDRUserField actually works?
Hi all,
I'm trying to add custom billing info into the CDR records. I did a
SetCDRUserField from my agi. Asterisk seems to acknowledge the call, but
the value is not anywhere in the CDR record. I checked the CSV and Mysql
CDR table. The field is always blank.
Anyone had any success with this?
Cheers and TIA.
2004 May 05
0
Missing: Red alarm event in Asterisk Manager Interface
Hi all,
This afternoon, while mucking around the manager API, I discovered that
my client doesn't receive red alarm events (i.e. when I unplug the
line). Astman won't work as well. I did made sure the user I logged in
as has read/write permissions for System events. A quick look in
chan_zap.c shows that when the alarm triggers or clears, Asterisk is
supposed to send a manager event
2004 Sep 16
0
problem connecting to icallglobe
Anyone has successfully used Asterisk with icallglobe's SIP termination
service? I'd been trying to get my Asterisk box to terminate
international calls through them. Asterisk seems to register OK, but
whenever I send a call to icallglobe's gateway, I always get a '403
forbidden'.
What I've done so far:
a. register with their gw in sip.conf
b. Defined a peer
2005 Mar 12
1
ipvolution TDM cards - vaporware?
Has anyone on this list gotten hold of these cards? It's been 2 months
since their official ship date.
Even the website www.ipvolution.com is in wee-wee land.
/leo
2004 Aug 10
2
WiFi phone radiation regulation?
All,
I just had the fortune to take one of the new Senao Wifi SIP phones for
a short test drive. First look - it's a nice, compact phone. Weighs
around 87g and roughly the size of a Nokia 6210. More on the those
later. The thing that struck me was the RF power, it's rated at 100mw
(20dBm). That's 10 times more than any of the other brands out on the
market Cisco, WiSIP, Zyxel
2004 May 18
2
My TDM-400P FXO experience
A bit about my experience with the TDM-04 FXO. Only saw a few post on
this subject, thought I would contribute a little about my experience to
save others the hassle.
a. As an earlier poster noted, the driver for the FXO is in the wcfxs
module. Perhaps it should be renamed to something less confusing.
b. You need the zaptel,zapata libraries from the cvs, the ones with
Asterisk 0.7.2 won't
2004 Sep 20
2
Update: Welltech Wellgate 3504A registration problem
Recently, there're some posts about registration problems with this
gateway. These are my observations using the latest firmware version 107a.
a) The gateway will register all 4 ports if you're not using password.
b) If using password, the gateway will only register the 1st port correctly.
For the record, the 2-port 3502 doesn't exhibit this behavior.
FYI.
2004 Sep 01
2
zaphfc crashes Linux
Hi all,
I'm having serious problem getting zaphfc to work on my box. I d/l'd
bri-stuff-0.1.0RC3/RC4a and followed the instructions to the dot.
Everything builds fine. But, when 'make load' the whole machine will
freeze. Anyone had the same problem and managed to solve it? I'm using a
Billion HFC PCI card on Trustix 2.0 running kernel 2.4.26.
As a side note, I feel that