similar to: Can't dial "Free World Dialup"--Loop Detected

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

Hola a todos...Auxilio, estoy varado en esto y creo que mi terquedad no me deja avanzar ... terquedad ya que estoy usando la estructura data.table y no logro obtener el resultado. Tengo una DT contiene un DATO y los resultados de la aplicación de una validación (ERR01, ERR2, ERR3) y la concatenación de estos errores en una sola columna. Si tiene error, graba el número del error, si no lo tiene,

Signed-off-by: ZhiHu <huzhi001 at 208suo.com> --- .../gpu/drm/nouveau/nvkm/engine/fifo/gk104.c | 40 ++++++++++++++----- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/drivers/gpu/drm/nouveau/nvkm/engine/fifo/gk104.c b/drivers/gpu/drm/nouveau/nvkm/engine/fifo/gk104.c index d8a4d773a58c..b99e0a7c96bb 100644 --- a/drivers/gpu/drm/nouveau/nvkm/engine/fifo/gk104.c +++

I'm trying to understand IAX and whether or not it would solve my difficulties: 'The primary goals for IAX were to minimize bandwidth used in media transmissions, with particular attention drawn to control and individual voice calls, and to provide native support for NAT (Network Address Translation) transparency. Another goal is to be easy to use behind firewalls.'

FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I think it will fail but we shall see. I really don't get the nerve of them (Free World Dialup has changed it's name to FWD) to ask for free ideas and development on a non-free service. Maybe if they can come up with a killer app and people will adopt it, then it might work, but then again, people

This may be slight off topic, but perhaps it has relevance: My iconnecthere account no longer works for "inbound" calls through NAT using the standard configuration that they provide on their website. I have sent them a message, but I believe it will be flushed down the toilet by the first-tier support people. When I call my iconnect number, it goes directly to voicemail. There

Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is

I'm currently using the SJPhone softphone with asterisk for remote SIP. When I dial into the voicemail, and attempt to pass the extension, I "hear" the sounds, but asterisk is not receiving any DTMF signals. If I use the Estera softphone, asterisk does receive the DTMF signals. Normally, I'd just say "Use the Estera" softphone to myself, but that's not an option,

Just change the install prefix directory in asterisk makefile to something like /usr/local/asterisk, and then you will see all the files needed for asterisk to run. > -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 22:03 > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] How could I install

Can anyone recommend a client / phone that runs on Windows XP, with either a sound card or some other hardware? Ideally free, but does not have to be. Thanks...

Hi everybody. I have to admin two offices, about 500 km apart. Because we're in Germany and dedicated lines (and even VPN ones over the internet) are very heavy in cost, we use a dialup ISDN connection. That is online almost 8 hours a day, but still cheaper. Now the remote office said that their e-mail was too slow and I installed a second samba server for them, being a logon master so

Dave wrote: > Hello, > I've got two streams, one for broadband, one for dialup. Well, having had > occation to use a dialup connection recently i checked the dialup stream. > Although it was streaming what the broadband stream was, the audio quality > was audibly worse. It didn't buffer, but it didn't sound as clear as the > broadband stream. This is expected.

Hello, I was wondering if it was easy to setup a dialup internet account system in Asterisk somehow(like PPP) laptop with modem -> PSTN line -> Asterisk server -> Server's broadband internet connection Has anyone done this? Thanks, MATT---

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

Hellow, * I have a problem with dial up signalling. currently I have configured asterisk server and E1 card to ISP. then other side I am having ATA to PC for connecting internet through DialUP connection. is it possible and please send me the procedure how I can do it ?? * ISP< <-> Asterisk <-> ATA <-> DIALUP -- Thanks & Regards, Vidura Senadeera, Sri Lanka.

This doesn't specifically answer your question, but is it possible for you to use an ogg stream? I've been running a stream using a 48Kbps ogg (i'm sure I could push it lower too...) and it sounds to me and other people pretty much just as "nice" as a 128Kbps mp3. We can't notice it being audibly worse. -Daniel On 12/30/05, Dave <dmehler26@woh.rr.com> wrote: >