search for: txlink

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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 __________________________________ Yahoo! Mail Mobile 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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
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