search for: supertec

...ng -- "Vendor # 701534" in their system) Hell, I was planning to get off their service anyways, if they would have allowed me time to properly port out the numbers, they would not have created an enemy for life. ---------- Forwarded message ---------- From: Rehan Allah Wala <rehan at supertec.com> Date: Sat, Jan 10, 2009 at 13:56 Subject: Your DIDX account To: Andrew Joakimsen <joakimsen at gmail.com> Cc: "muneeb @ supertec. com" <muneeb at supertec.com>, suzanne at supertec.com Thank You for this email Andrew, Please move your numbers in next 3 days somewher...

...t from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already tested: Xpro Logged in on Machine B using ILBC sending to Machine C and it works fine. Do send me your charges. Thank You, Rehan Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today!

Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and

atlast after working of 7 hours i got voange soft account working on asterisk.

...hat comes from the mailing list or the didx server? > > IF u can forward the didx email then i can check that > > Rehan > > Date sent: Fri, 19 Oct 2007 10:23:00 -0400 > From: Steve Totaro <stotaro at first-notification.com> > To: rehan at supertec.com, > Commercial and Business-Oriented Asterisk Discussion <asterisk- > biz at lists.digium.com> > Subject: Re: [asterisk-biz] DIDX Receives Digium Innovation Award > > >> Rehan, >> >> Pleas fix the time on your email server. I do not need...

On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala <rehan at supertec.com> wrote: > Your service is still up and working, Because Suzanne Bowen has better judgment than you. > You did charge back on the payment to us, That is correct. There is $86 balance in my account I did not expect to get back by just asking for it. > We are being nice to you an...

...Would like to be able to offer residential users the ability to destroy their dialplans (remote call forwarding, find me follow me, see/print bill) etc. I'd appreciate real world experiences and have already been down the VoIP-Info road looking at the majority of platforms offered (A2Billing, Supertec, etc.) with no success. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP "Enough research will tend to support your conclusions." - Arthur Bloch "A conclusion is the place where you got tired of thinking" - Arthur Blo...

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

Dear Asterisk Fans, I'm an Asterisk consultant in Argentina and want to make an spanish wiki (something like voip-info.org). I have the idea and some concepts about this project. It won't be a comercial project, it would be free and it's target would be spanish talking asterisk enthusiasts. I'm wrinting these for the sake of finding contributors, people who want to help me

Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo "${UNIQUEID} =>" >> /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around.... much thanks, Paul Hales

Message: 20 Date: Mon, 16 Jan 2006 00:18:18 +0000 From: Mike Hemstock <mike@csits.net> Subject: [Asterisk-Users] Choosing an FXO card To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <200601160018.18517.mike@csits.net> Content-Type: text/plain; charset="iso-8859-1" Folks, I'm looking at buying an FXO card

Dear All, I am using Asterisk CVS-HEAD-05/05/05-15:46:09, and found it using up 99% of cpu 6-8 times in a day, even when its doing nothing or even if it is, its not supposed to eat all the cpu. whenever this happens, I need to stop it by issueing "stop now" 15-20 times on CLI or kill the process, then start over. Please suggest what to do. regards, Umair bari

Hello All, i m having busy signal when i dial any number, while incoming on zap is working fine and its transfering to my soft phone. some time back outgoing was working ok but now i dont know what i messed up. any idea ? it gives busy signal after Zap/25-1 answered SIP/300.... -Neo = Spawn extension (voicepulse-incoming, s, 1) exited non-zero on 'Zap/25-1' -- Hungup

Dear All, For those who celebrate Eid. I would like to wish you a very Happy Eid Mubarak. For those who do not know what it is, Its a Prayer in memory of Ishmael son of Abraham, when he attempted to sacrifice his beloved son on request by god. Muslim's celebrate it with a sacrifice of a goat every year. I belive Christians & Jew's belive in the same. Peace and Harmony to all.

It looks like you have the first extry of the [incoming] context in extensions.conf commented out Hi all, I am resending this message, so far no one has helped me with this incoming call issue. there is no problem with outbound call but there is no inbound call to my sip phone. the only message I get when I call from pstn is "unable to create local channel for call forward to

I would like to make a suggestion and recommend that you put your Asterisk box on the outside and let it also pull duty as your firewall/nat router. The iptables overhead will be minimal on the system and you'll save yourself a lot of headaches in the long run. The biggest problem being that having an asterisk server behind a nat, and then also having sip phones trying to connect to said

I have a DID vendor that wants me to be able to generate specific SIP error messages under certain conditions and I'm completely stumped on how to do these: #1 - They want to see a SIP 503 error response(service unavailable) when they send the call in to an active extension and and the service is not available, I don't have a clue on how to simulate this. #2 - When they send in a

Hi Everyone, I am dealing with a problem for now and its really annoying. I want to hangup calls from AGI but it seems that my AGI is not rejecting the calls properly. { $agi->verbose("number-not-in-service"); $agi->exec("Congestion","1"); $agi->hangup(); exit; } with the above logic, all of

I thought the same thing before I made my reply but zapata.conf seems to be the only config file that deals with echo at all. From what I understand of 'echotrain' is that at the beginning of the call it sends a short signal out that measure echo in an attempt to try and cancel it. I was wondering if you tried using it and if so was it of any help? Sincerely, Steve But is Zapata.conf