similar to: Realtime IAX

Whoa, talk about flying under the radar. I got a few DIDs *months* ago from Sixtel (or iax.cc). Initial respone was great, but then it seemed that the only tech-support person had fled the country. No responses, bad responses, poor call-quality. I had a few $s left in the balance and kind-a just forgot about them. Not worth 10 minutes of my time to get $10 of my money back, quite frankly.

hello list, I subscribe to Sixtel.net for a DID just to see how it worked. they say DID active inmediatly , but after 4 days , I have no DID , I tried to call tech support office and it seems they all ways are in hollidays, a support ticket no good luck, I call back service also not good luck. so anyone here has experience with them? are they a SCAM? fortunally only put there $10 bucks but

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

Hi I looking a good IAX service for a *emerging * voip provider. Better with a test account to try. Thanks in advance. roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte t?cnico ISPs Jabber ID: rpereyra@lugmen.org.ar For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989 -------------- next part

I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening: localhost*CLI> -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtest May 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019

Is iaxtel down? Ive been getting this: May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest: Auto-congesting call due to slow response -- IAX2/Iaxtel-12 is circuit-busy -- Hungup 'IAX2/Iaxtel-12' is it down or am I doing something wrong?

The past two days, I've been having issues with my two VoIP service providers where calls just suddenly hang up. The following is from the log: Mar 14 13:50:55 WARNING[5887] chan_iax2.c: Max retries exceeded to host 64.34.45.100 on IAX2/voipjet-3 (type = 6, subclass = 11, ts=250000, seqno=80) Mar 14 13:50:55 DEBUG[10428] channel.c: Didn't get a frame from channel: IAX2/voipjet-3 Mar

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

Hi, I've tried to make toll-free DID work for the last 2-3 weeks. Apparently only the IAX.CC/Sixtel personnel can make a call to my toll-free. Anybody else just gets a busy signal. It takes for them about 5-6 business days to respond to my request. It seems they are looking in the once a week to the High priority tickets. It also seems that they are permanently closed ! No matter when I

I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel:

Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD

I apoligize in advance for this newbie question on what I perceive as a mostly advanced level list... I did some searching, but would like some of your expert opinions. I'm building an asterisk server to be used in the home, both to learn, and as proof of concept of applying this solution in a home. To keep costs down, I'm considering one X100P ($25 ebay clone) card to connect the

If anybody here is a sixTel customer, can you share any tips & tricks for getting in touch with anybody there? They are absurdly hard to get a hold of, particularly when you have a technical issue needing to be resolved. If anyone has any phone numbers other than their main 800 line, I'd sure appreciate it. Thank you, Bryan -------------- next part -------------- An HTML attachment was

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US

Hi List, I have set up the following in my extensions.conf ; local numbers look like 0262XXXXXX ; but must be dialed 262 262XXXXXX exten => _0262XXXXXX,1,Dial,IAX2/543@voipjet/011262262${EXTEN:4} exten => _0262XXXXXX,2,Dial,IAX2/jhiver@NuFone/011262262${EXTEN:4} exten => _0262XXXXXX,3,Congestion It did work for a while, however when dialing I get: stargate*CLI> -- Executing

Hi Everyone, I have a DID number with livevoip, but I have been experiencing two problems that I can't seem to resolve. I am not sure if they are in any way related. I have other DIDs with iax sixtel but I do not have that problem. Livevoip seem to think that the problem might be with my configuration. Can someone help me figure out this problem please. 1) When an incoming call to my

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

I was just checking out the dids for all of my fail over providers and noticed that neither DID that I have with SixTel work. Both pause for a long long time The local number gives a recording: 'The number you have dialed is not in service or is assigned in a different area code. Please check your number and dial again'. The 800 number just rings busy. Anyone else having this issue or

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the