similar to: Voip* 300 minutes limit, credit expires

=20 Andrew Fenn wrote: > You don't need their program to use justvoip, voipdiscount, etc=2E You > can use any sip client to connect to Betamax servers=2E Try Twinkle=2E >=20 > On Mon, Jul 27, 2009 at 11:24 PM, miroa84<wineforum-user at winehq=2Eorg> wrote: >=20 > > I tried to install justvoip several times and I cannot install it=2E Can somebody tell me how to

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to

Hi, sorry for this a bit OT. I'm using VoiceTrading for some calls -premium route- and can't get CID to work despite the fact that CALLERID(num) and CALLERID(name) are setted. I ask in VT->myAccount to accept calls from my IP without checking username & secret. On incoming calls the CID is setted to 0100000000 If I accept calls from username & secret and no IP relation,

I tried to install justvoip several times and I cannot install it. Can somebody tell me how to install it on ubuntu? Meybe next version of WINE will support it?

It won't be long until big labels attempt to eliminate the digital ins and outs of equipment. The problem they face is most people that currently use a rack system featuring digital interconnects will NEVER revert to an analog only system. I know that I won't! Think back to VHS, BetaMax, SVHS and LaserDisc. VHS has incredible market share because its licensing is open. Betamax was

I am sure, after I have it setup once, everything will be cristal clear. I could not find a documentation, maybe together we can make at least a "README" for the next user ;-) 1. Brands: Brand Name Create a brand name you want to market your pre-paid card Language Choose one of the available languages en, de, fr, es Published Number ??? DID ??? Guess:

Hello, I have a Sip phone (Siemens C470IP) which works perfectly with different VoIP providers (iptel, betamax, ovh...). It also worked well with my testing server (ubuntu and inside the LAN). But now the problem i have is that the hardphone doesn't connect to my dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing is that ekiga can connect to the same asterisk server with

Andrew Fenn wrote: > > If you're trying to just phone people using the voip service then you > can use any sip client to do so. Twinkle and Ekiga are two such linux > applications you might want to try instead of the one provided by > Betamax. yes i now, but i use voipcheap because it allow to connect my home phone with my iterlocutor phone directly without headsets or network

On Tue, Jan 17, 2012 at 09:44:25AM +0100, rs at noveltech.de wrote: > We are currently try to add Gapless support on our device? If we rip an CD > with our device, we can find out, that one track follow after another so we > can > recognize, that the tracks are gapless or not. Are you aiming for maximum compatibility with how the source CD would be played on an average CD audio

Hi all, did not find much info in lists about subj. I have ztdummy working properly because I can use conferences without any errors. But when I try to use trunk=yes, I get the following: Sep 2 21:20:51 WARNING[1137720112]: chan_iax2.c:6422 build_user: Unable to support trunking on user home' without zaptel timing Sep 2 21:20:51 WARNING[1137720112]: chan_iax2.c:6246 build_peer: Unable to

I am still looking for a solution and I am sure that I am not the only one having that problem: If provider A fails for any reason, the next provider should be taken. There are many reasons, why a provider fails, like: password wrong (cli reports so, but actually it is the gateway's problem) gateway temporary not reachable gateway busy ... Our user places a call, the gateway responds with

>Our user places a call, the gateway responds with no sound at all, or >hangs up, or gives busy tone. > >How can we get to the next provider? > >I have now: >exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-a) >;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-b) >;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-c) >exten =>

I try to connect to voipjet, but I get always busy, ... it worked yesterday, ... no changes on my side.... -- Executing SetGroup("SIP/615-829b", "iax-voipjet") in new stack -- Executing Dial("SIP/615-829b", "IAX2/17567@voipjet/011886228357765") in new stack May 20 18:16:26 NOTICE[9733]: app_dial.c:973 dial_exec_full: Unable to create channel

Polycom 601, hands down. - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Warren Sent: Thursday, June 15, 2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk? If you had approx $300 per phone as a budget and needed to buy

Hi i'm trying to install this application by wine installation has been successful but software doesn't properly it doesn't start at all or it's giving me critical error msg

Adam, I thought Andrew Kohlsmith gave the individual good advice without intentionally malaciously spitting in the guys face. For the question, " 'Whats the ' Minimum CPU required for 60 calls? " I think a Pentium 3, high end, which is cheap right now, should do fine, but you will need either 3 T-1s or arrange for the calls to come in via SIP, but you will still need more

All we have a deal on Canadian termination. Rate: $0.0039 US Dollars Billing: 1/1 Protocol: SIP or H323 Codec: G729 Terms: Prepaid Only. We have a real-time web interface where you can monitor or download your CDR's. Please e-mail me offlist if you are interested: jweisman@ibell.net Thanks, Jon -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've compiled and start the modified prepaid with postgresql. I would like to ask if anyone can give me a sample account to be populate in the database. I also want to confirm if this correct. I created a database prepaid (createdb prepaid) and from the asterisk-prepaid(current modified prepaid application) database folder, I execute (psql prepaid < prepaid.sql, psql prepaid <

Prepaid app can not connect to the database, [app_prepaid.so] => (Prepaid Application) == Parsing '/etc/asterisk/prepaid.conf': Found Jun 22 14:38:43 ERROR[-1084964736]: app_prepaid.c:127 check_connected: app_prepaid: cannot connect to database server localhost. Calls will not be logged == Registered application 'Prepaid' Here is pgsql confs; pg_hba.conf: ------------