similar to: Beware of DIDX & Super Technologies

All of the emails I get from the list have the correct time with the exception of the typical 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

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 and you do not understand the

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending 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

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

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

I'm getting: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00009ms SCall: 00008 DCall: 00002 [66.98.180.77:4569] CAUSE : No authority found Under the DIDx number I'm putting IAX: default@asterisk.crystalstream.net/1567252NNNN(IAX) where N is the rest of the numbers. In iax.conf I have: [default] ; DIDx type=user

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for

Hey anyone know once you've bought a DID from DIDx under the Purchased DIDs tab you can click a link for one of your numbers that says '0 (SIP)' and when you click that like there's SIP and a space or the URL and AIX and a space for the URL.... What do I put here? How do I setup Asterisk to accept these Joshua __________________________________________________ Do You Yahoo!?

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:

Hi Friends, We have created a video based training for Asterisk in English and Urdu. Please check them and let us know how we can improve them for no-voice users. http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU

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

I am using DIDx.net as my DID provider but they don't seem to get their act together. A lot of times the phone numbers don't work. How can provide my own DID, my asterisk server is being hosted at a Data center and has a reliable vendor that does my termination and do SIP to SIP and have no T1 channels. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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.

Don't know if anyone's got experiences on this they'd be able to share... I'm trying to obtain numbers for Kenya/Nigeria, but I'm struggling to find a company selling them. There's one for Nigeria listed on didx.com, but $22/month seems a little steep. Has anyone had any luck getting DIDs for countries in Africa, and are there any companies out there selling them? Thanks

Rather than requiring the scopes to be the first members of their respective structures (so that casts can be used to switch between the different views), properly use types and container_of(). Signed-off-by: Jan Beulich <jbeulich@suse.com> --- a/xen/drivers/passthrough/vtd/dmar.c +++ b/xen/drivers/passthrough/vtd/dmar.c @@ -304,13 +304,15 @@ static int __init scope_device_count(con

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

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

On Wed, Nov 28, 2018 at 05:55:44PM +0200, Mika Westerberg wrote: > On Wed, Nov 28, 2018 at 10:09:22AM -0500, Michael S. Tsirkin wrote: > > Yea all this is weird, in particular I wonder why does everyone > > using dsm insists on saying Arg4 > > when they actually mean Arg3. ACPI numbers arguments from 0. > > > > So it's a bit ugly, and maybe worth fixing but

Hey all, I have a couple of questions. 1) What is the maximum amount of registrations and ongoing calls you've been able to achieve on your Asterisk systems. Please do not respond with marketing hyperbole. I'm looking for real world implementation in the thousands range. For instance, max I recall having on one souped up server was a couple hundred registrations with no more than 70

Not sure if it is just my configuration but building the RPM on Centos x86_64 fails. Also not sure if this is to be fixed in the Makefile or in the configure. The rpm compile proceeds apace until the build reaches the %files section. Here it uses the _libdir directive to look for the libraries that have just been compiled. However in an x86_64 environment the _libdir directive points to lib64