similar to: A New Phone Service - www.virtualphoneline.com

I assume most people here know what a joke DIDX is -- but in case you didn't already know, please avoid these people. Basic features of their service don't work, their tech support refuses/drags their feet to fix them for a month and if you post publicly about them, they terminate your service. Instead of investing their effort in reading mailinglists to terminate customers maybe they

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 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. > We are being nice to you and you do not understand the

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:

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

The US has a Do Not Call list to which people can subscribe to prevent being called by advertisers. Federal laws (strengthened by some state and more local laws) assign penalties for calling people/phones on the DNCL. Is there a query gateway that Asterisk (or an app using Asterisk) can filter through to ensure a number is OK to call (not on the list) before calling it? -- (C) Matthew Rubenstein

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 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.

Hi, I recently purchased two X100Ps and a TMD40B. I have set everything up according to the instructions however I can't get asterisk to pick up an incoming call (It just keeps ringing). Also, when I pick up an internal extension I get dead air. I searched the mailing list for 3+ hours -- What am I missing? Thanks in advance for your help. Here are the installion steps that I used:

I am looking for someone that could share their code for this function: Outgoing call -> macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones -> if human connect to agent else hangup and write status to cdr. Need help with this! Regards / Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Please, People, help me ! When I start the smb services, I receive this message: [root @g 11:25:39 /root]# service smb restart Stopping service SMB: [ OK ] Stopping service NMB: [ OK ] rm: cannot remove `/var/lock/subsys/smb': IO Error Starting service SMB: [ OK ] Starting service NMB:

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

A friend of mine recommended checking out the Vyatta project. I was reviewing their website and saw endorsements of their product running on both Vmware and Xen. In fact, they have a Vmware virtual image and a beta XenServer image. However, I am running a typical Debian Lenny 64 DomU installed through the repositories. According to their installation documents, it states that the ISO can be

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

Someone got ahold of my information (actually, I know who they are - t2net.net - just not how to get back at them) and now I'm getting 4 phone calls a week upwards of 20 spams a day. I'm trying to put an end to the spam at least and the built-in stuff for Evolution isn't working as well as it should I did some Googling on procmail and fetchmail and spamassassin config files and much

Hi All, I read the advanced networking howto and find a useful way to shape my adsl link. is says: "tc qdisc add dev ppp0 root tbf rate 220kbit latency 50ms burst 1500 [...] Change 220kbit to your uplinks *actual* speed, minus a few percent. If you have a really fast modem, raise ''burst'' a bit. " My adsl''s upload brandwidth is about 120 kbits/sec,

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of empire underground Sent: Friday, January 09, 2004 1:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Screen Pop & Remote Agents > can I put a .csv file in the sql DB and have it dial from there? and will I be able to set a > Dial Plan to

Hi, Not sure where to address this question ... but I guess this is a good start. I am a freelance Rails developer - and have been developing Rails apps professionally for about a year now. I just want to get an idea of what you other freelancers charge yourselves out at. I have been charging the equivalent of about $28/hour, but get a sense that I could/should charge a whole lot more. I am