Displaying 20 results from an estimated 30 matches for "txlink".
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tlink
2004 Jul 23
4
hang up when going to voicemail
...e over IAX. That works. However, if caller hits 4 to go
into voicemail, the system hangs up. Am I doing something wrong in the dial
plan, or is this a CVS change? I had no trouble with this until I upgraded
to about 07/21 CVS, and I'm on 07/23 [latest] now with same results.
My dial plan:
[txlink]
exten => s,1,Answer
exten => s,2,Background(/txlink/txlink-main)
exten => 1,1,Dial(IAX2/####:####@####/12149490280)
exten => 1,2,Hangup
exten => 2,1,Dial(IAX2/####:####@####/14693373687)
exten => 2,2,Hangup
exten => 3,1,Dial(IAX2/####:####@####/18174017579)
exten => 3,2,Han...
2005 Oct 18
8
free dids on goiax.com
...se down let me know. The
best ideas I have now is to only allow a certain amount of calling per
month, add velocity checking, and somehow put some accountability into
the sign up process to keep the prank callers and multiple account
abusers away.
yours,
Matthew Simpson
GoIAX -- www.goiax.com
TxLink -- www.txlink.net
2005 Sep 23
4
goiax expanded with free us domestic calling
I launched www.goiax.com last week, which is intended to promote the use
of IAX as a free and open source alternative to products like skype.
There is no charge for the service. Right now I have free outbound to
united states toll-free and us domestic numbers working.
Currently the site hands out a virtual 87820-xxxxxxx number but I intend
to add the ability to get a free United States DID
2005 May 19
1
(no subject)
...almost, that $25,000 per month bandwidth cost to me.
So if Digiums DS3 Channelized Voice PCI card costs,
around what Sangomas costs, $6,000, (JUST AS A EXAMPLE
FOR THIS POST), $12,000 for 2 Digium DS3's in 1 month,
I will save almost $10,000 AUTOMATICALLY and ever
month thereafter! :)
Come on Txlink DID #'s.
Come on Digium with the DS3 Channelized Voice PCI
card.
Then all Digium would have left to do is create a
board
or work with someone on getting Radio Waves into your
computer. :)
Sincerely,
SoftwareRadioGuy
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Take Yahoo! M...
2005 May 19
1
Do Both! :) Re: Telecom SIP termination vs. DS3
Message: 16
Date: Thu, 19 May 2005 00:16:34 -0600
Michael,
Do both!
As for Sip Termination:
-----------------------
Contact Kristi Eggers @ Txlink.net for month to month
Originating/Termination VoIP Toll Free or Local USA
DID #s. Yes they do both Sip and IAX. You must have
seperate accounts for either Sip or IAX and fund your
account with a minimum of $100. This is what I did.
Once I get through testing out my Asterisk/Areski
Calling Car...
2004 Jun 01
0
free sip termination
help me test load a box!
I have a new box with four PRIs on a TE405P
I will terminate US Toll-free traffic (1-800, 888, 877, 866) for free via
SIP to anyone who wants to test. Just email me at matthew@txlink.net if you
would, to let me know that you're testing, and with any comments about
quality,
etc.
I have ulaw, alaw, and GSM codecs enabled.
To use, just send your call via SIP to 67.153.209.214 with the username of
"free" secret "free"
yours,
Matthew Simpson
TxLink Communi...
2004 Jul 02
0
DISA and AGI: authenticate by caller ID? (resolved)
...code to snip everything but
the 10 digits. Adjust accordingly if you have more or less
than 10 digits. Also, I've thought of a bug already, if your caller ID name
has digits in it, it'll break the regexp. Adjust accordingly
if that is true about your installation.
Yours,
Matthew Simpson
TxLink Communications
IAX/SIP Termination and Origination
Wholesale Dialup Services
matthew@txlink.net
972-617-2877
http://www.txlink.net
You'll need a context called ldincoming [or equivalent] for the AGI to
transfer access to DISA like:
[ldincoming]
exten => 1011,1,DISA(no-password|disa)
exten...
2005 Mar 06
1
Re: [Asterisk-biz] Livevoip U.S. 800 LNP Starts March 9th 2005
...they have not. Calls are failing again today. They have offered to
refund my money but that does not solve the problem. My asterisk server
is only 4 to 12 ms away from their "network". I have had VERY good luck
with nufone.(40 to 45ms away) Only have 1 or 2% fail rate. Going to be
calling txlink.net on Monday.
Seems that LiveVoIP does not care about asterisk users. They like to
pass the blame.
-Tim
On Sun, 2005-03-06 at 17:04, Mike Dent wrote:
> Hmmm, I was contemplating going with livevoip, glad I read your post.
> I'd be interested if they resolved your issues?
>
>...
2004 Sep 19
6
new ATA box for sale by Linksys
Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99
retail. They have a version with a router for $89.99. We picked the
non-router version up and it seems to be a rebadged Sipura SPA-2000. The
box has a Vonage service package inside as well, but it does work with other
services.
The box also has a "User Guide" meant for end-users that is very well
written [no
2005 Jan 10
0
TE-405P freezing, anyone else?
...the same issue and that he
had heard that it was an issue with the firmware on the "newer TE-4x0P
cards".
BTW, I did a trace on what the card was sending with a protocol analyzer...
card was sending out all "1"s.....
A new TE-405P seems to be working okay.
Matthew Simpson
TxLink Communications
http://www.txlink.net/
+ SIP and IAX origination and termination
+ Unlimited incoming toll-free $20/LATA
+ Texas origination and termination for $0.005/min
+ US origination and termination $0.005 to $0.012/min
2005 Feb 03
2
Good 800 Number provider
--On Thursday, February 03, 2005 2:20 PM -0500 Andrew Thompson
<asteriskuser@aktzero.com> wrote:
> What you are seeing with these bargain providers is they have a clause in
> their contract that says they own the number, not you. It is a lock, and
> it ought to be illegal, but sadly, it's probably not. If you choose one
> of these companies that doesn't allow you to
2004 Oct 04
3
motherboard for T100P
anyone have a recommendation for a place I can buy cheap motherboards that
supports those 64-bit 3.3 volt PCI slots for the T100P ?
I can't find them at Fry's or anywhere locally. All I can find online is
dual processor server boards that are overkill for this application.
I would like to use a P3/ P4/ AMD single processor. No Xeons or dual
processor junk.
Anyone know why digium
2004 Jul 28
3
faxing
What are your experiences with faxing through Asterisk to the PSTN?
We are using g.711u as a codec, and are originating/terminating with Broadvox as
well as through our own PSTN gateways.
We have had some luck with incoming faxes coming into our network from Broadvox
DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody
that is using FAX is in our local rate centers.
2004 May 28
4
Wiki TOS - worrying for an open source project?
Hi there,
I've made a couple of small contributions to the wiki but recently I
read the Terms of service, they are pretty draconian:
LICENSE AND SITE ACCESS
voip-info.org grants you a limited license to access and make personal
use of this site. This license does not include any resale or
commercial use of this site or its contents. Without express written
consent of voip-info.org you may
2004 Jul 28
1
is chan_skinny broken?
I am trying to use chan_skinny but when loading the module I get:
[ Booting....../usr/lib/asterisk/modules/chan_skinny.so: undefined symbol:
ast_pickup_call
I am using CVS 07/23
I can't get chan_sccp2 to compile, it gives me parse errors, or I'd be using
that. :-/
2005 Jan 19
1
ztdummy issues on new asterisk install
...2113 1 zaptel
I did 'service start zaptel' and everything appears cool.
However, in *, I keep getting the messages:
Jan 19 15:29:53 WARNING[15818]: Unable to open IAX timing interface: No
such file or directory
Jan 19 15:29:53 WARNING[15818]: Unable to support trunking on peer
'txlink' without zaptel timing
I deleted /etc/zaptel.conf (as was recommended when I searched the
lists), but * still won't recognize ztdummy.
Any hints or advice would be greatly appreciated. I'm probably just
forgetting something stupid...
Thanks again,
Jake
2005 May 19
2
cisco 7960 question
I have a stupid question. How do you remove line presentations on a cisco
7960 ? I have 3 line presentations I don't use anymore and I can't figure
out how to remove them.
2005 May 20
0
ref: Cisco 7960 question
Message: 5
Date: Thu, 19 May 2005 21:44:11 -0500
From: "Matthew Simpson" <matthew@txlink.net>
Subject: [Asterisk-Users] cisco 7960 question
To: <asterisk-users@lists.digium.com>
I have a stupid question. How do you remove line presentations on a cisco
7960 ? I have 3 line presentations I don't use anymore and I can't figure
out how to remove them.
If you look in...
2005 May 24
0
Re: origination providers (mike castleman)
...e back, or want me to sign an NDA just to
>get a rate quote, or some other bullshit. Most of the
>providers whose rates are plainly posted on their
>website have a limit of at most 4 or 6 simultaneous
>calls, which is not likely to be enough for the
>application I'm considering.
Txlink.net, ask to speak with Kristi Eggers.
>You can reply off-list or on-list, as you prefer.
>many thanks,
>Mike
Martin
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2005 Jan 21
1
problem with TE-405P
Hello, I have two TE-405Ps that I am having trouble with.
I'm using an Intel 865 motherboard with a Celeron D processor. Kernel 2.4.26,
Slackware 10.0.
my /proc/interrupts:
CPU0
0: 172317 XT-PIC timer
1: 2 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 2003 XT-PIC eth0
8: 1 XT-PIC rtc