similar to: sip call through dialup connection

Hi, Currently streaming ogg isn't practical in this situation. That was one of the first things i checked into. WHen i looked i didn't see a streamer that did both ogg and mp3. Thanks. Dave. ----- Original Message ----- From: "Daniel Ballenger" <lpmusix@gmail.com> To: "Dave" <dmehler26@woh.rr.com> Cc: <icecast@xiph.org> Sent: Friday, December

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. I used lame to encode the tracks to mp3 and used it's

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---

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

Hello All, I have a number of X-Lite users in countries where the incumbent Telco will do anything to block VOIP traffic. For some reason neither the X-Lite broadband or dialup clients would register with my server unless we configure them to use the Xten Xtunnels demo server. Once the Client has registered the call quality is great! The problem is the Xtunnels does not support other IP Hard

i can register to asterisk with msn, place calls, etc. However, if I = unplug the ethernet cable and get a dialup connection I get the = following error when i try to place a call with MSN (i can register = withoug problems); chan_sip.c (process_sdp) No compatible codec! I have disallow gsm and allow ulaw on my sip.conf Any help would be greatly appreciated. Dan

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.

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

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: >

This is another reason aacPlus is an advantage. It is supported using Orban Opticodec-PC Streaming Encoder, Icecast2 Servers and Winamp. http://www.orban.com/orban/products/stream/1010_overview.html Icecast2 Eval Stream here: http://www.orban.com/orban/products/stream/orban_eval_ice.pls -greg. At 11:05 2005-12-30, Dave wrote: >Hello, > I've got two streams, one for broadband,

Hi, I'm away at a conference in Amsterdam. My home is in Cambridge in the UK. On a whim, I tossed an ATA186 and a phone into my bags before leaving home. I was able to plug my ATA186 into a LAN here at the conference and was connected to my home Asterisk in a few seconds. Total time from unzipping my bag to talking to home no more than 15 seconds. OK, so the kit could be more portable,

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.

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

Hi, I am new in asterisk. Is it possible to use it with common dialup modem to connect ptsn to the server? Thanks Regards, Soragan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040123/8e77b5cf/attachment.htm

What would be a good combination to use on dialup connections? I know iax is better than SIP, but I dont' know much of anything about the various codecs. Also, how well would an iax or sip solution work compared to skype as far as voice quality? If it helps, here is what I'm trying to do. We know about 30 different families in one area in Russia that have relatives here in the US.

Hello everybody, I've searched google for a while looking for a list of USB or PCI dialup modems supported by CentOS without success. Any information on this subject will be really appreciated. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20081125/dd8598ed/attachment-0003.html>

Hello, I've got a FreeBSD 6 machine running icecast2 and ices0streaming four different streams of mp3's. Now i have a need to stream the same content as either aac+ or ogg vorbis whichever would be easiest to set up and give the best quality. I want these new streams to handle both broadband and dialup users without sacrificing quality. My problem is i really really do not want to

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

Hi Folks, I'm sure I'm speaking for more than a few (though possibly a minority) here. There's something of a hidden assumption that R users can readily download whatever they need from CRAN. Some of us are on narrow bandwidth dialup connections, so downloading large quantities of stuff is out of the question (e.g. at approx. 5min/MB, it would take over 2 days to download a single