search for: rehan

...cal list slowness. All of your emails (and only your emails and spam) are approximately 11 or twelve hours in the future. The email I am responding to has the correct day but the time reads 11:13 PM. I am using Thunderbird 2.0.0.5. If using Outlook, I think the time is correct. Thanks, Steve Rehan Allah Wala wrote: > You mean the email that 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-notifica...

...ding to Machine C Machine B is an asterisk box, transcoding it 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!

...h them were from Global Crossing -- "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 you...

...r, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and http://www.virtualphoneline.com/did for current available numbers. Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129b&OID=100000 Let me know how it goes for you, Rehan Ahmed EmailID: 25

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

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....

...se 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. Rehan

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:

...ge: 5 Date: Fri, 14 Jan 2005 08:16:22 +0200 From: "Rob Lith" <rob@connection-telecom.com> Subject: [Asterisk-Users] RE: [Asterisk-biz] SS7 and Asterisk solution To: "'Commercial and Business-Oriented Asterisk Discussion'" <asterisk-biz@lists.digium.com>, <rehan1@rehan.com> Cc: asterisk-users@lists.digium.com Message-ID: <200501140616.j0E6GPag004475@aphrodite.dbuzz.net> Content-Type: text/plain; charset="us-ascii" Tracy, one example I can think of is here in South Africa, when VoIP is deregulated on the 1st February the very first trick...

...xt step is to get speex native on some hardphones (already hassled a Snom developer to look at putting it in their phones). I'll be releasing an Asterisk install script with optimised speex codecs (ie with sse optimisations) so it should go head to head with g.729 performance wise. Cheers Rehan Khan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20061207/bf05e8ad/attachment.htm

...ut please follow- up to the -biz list only. I have set the reply-to, but I fear mailman will strip it off ... Please don't flame me for posting to -users, I'm just not sure who lives on -biz as the (signal/noise | net.kook | troll) factor has been pretty bad on that list lately (poor Rehan!!) ;-) Telephony Depot (www.telephonydepot.com) is always looking for talented people who get excited about open telephony platforms such as Asterisk (in its various forms), FreeSwitch, Yate etc, and who just plain find telephony hardware strangely fascinating ;-) Despite the media's r...

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

Hello, We are looking for commercial solution SS7 with Asterisk. It does not need to be "build-in" with Asterisk. Could anybody suggest something? Thank you in advance. Bart