similar to: modem through Zaptel/Digium?

I've seen there are spandsp, iaxmodem to simulate a modem for receiving faxes under asterisk. Hylafax can be "tricked" to think it is a real modem. Can I create a dialup/access server with the same method? Can ppp run on top of this "software" modem?

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

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>

Dear All Please be informed that I checked for the presence of internal modem on my CentOS server , as the followings : #dmesg |grep -i modem #lspci |grep -i modem #lshw |grep -i modem According to the output , it seems that my CentOS client does not contain internal modem . So I decided to add external USB modem and make use of an PCAnyWhere like application that enables for Remote PC Access .

I've been a happy user of asterisk for over a year just for a small home setup (a Digium TDM400P with one POTS line and three internal extensions plus a couple of SIP phones). I recently moved from running Fedora Core 6 running * 1.4.1 compiled from source and zaptel 1.4.7 to Fedora 8, using the RPM package for * 1.4.18 and zaptel 1.4.9 . It worked fine for a week, but suddenly I can't

Hi All Im having problems setting up a dialup connect in CentOS PPC I get the following error when attempting to use wvdial (modem set as /dev/ttyS0) --> Cannot open /dev/ttyS0: input/output error Ive tried kppp but ive had not luck Anyone know how to fix this? normally I would use pppconfig and pon/poff but havent been able to use/find them, are they install by default? __ Note: this

On your web you have a link http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard To Setup Calling with Diamondcard.us and I signed up and paid the money according to Stephen Karrington it was all automated... And it was automated to take money but when you look for service hookups or information you don't get it. I have tried now for last little while to contact them for support

Rather off-topic, but I'm looking for IP-based KVMs (~16 ports) that can handle both PS/2 and USB hookups on the server side. All of the answers over at Slashdot are a few years out of date and it looks like prices on KVM head units have dropped a bit over the years. Some of the older units only worked with Windows, Internet Explorer and ActiveX. Others like the ATEN KH1516i supposedly

I have a friend with a PRI coming into a modem bank that is receiving 56K modem calls and some ISDN data calls. He wants to dump his analog office phone lines and use some of the capacity on the PRI. I have been digging through the mail archives and Wiki site on this subject but the information I found doesn't give me a high enough confidence to go buy a few T1 cards and try it out.

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---

Gentlemen We are attempting to use * in a simple switching application: +-----> office lines | V LEC <--PRI--> * <--PRI--> modem bank (56k dialup modems) The problem is that (even with no office lines active) the modems have difficulty establishing a connection, the connection is slow (way too slow for 56k modems) and the connections are

Hi, maybe this has been asked before but I couldnt find a proper answer on the web or list. I want to use a analog V.92 modem to make outgoing (and possibly) incoming phone call through a standard analog phone line. I found on web it's easy been done via chan_modem.so module. But this seems removed from asterisk or buggy. So my questions are can I enable chan_modem.so to be built or

Hi, I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall (all ports we're set according to instructions) on DSL line. When pbx connects to Digium demo server( I'm located in Slovenia, Europe), it gets first few words, then silence and then comes back when enumerating dial possibilities ("4 for accounting ...). Same happens from SIP or IAX local extension.

I use Speex with dialup modem users. Even if the dialup modem is "56K", you should assume that upstream bandwidth available is only 20-30kbps at the most. I use 16kHz wideband mode and VBR quality 2. Also, I send 80ms (4 frames) per packet, because there is an overhead of approximately 33 bytes per packet due to UDP (8 bytes), IP (20 bytes), and PPP (~5 bytes) headers. If you

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

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

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

Hi there Is it possible to configure ssh so that a session can continue/recover after a change of IP? I have a number of static IP dialup accounts and there is no problem using these because when my dialup connection timesout and later reconnects, it gets the same IP and so any existing ssh sessions are unaffected. However, these accounts with static IP's are metered (i.e. expensive ;-).

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

Good day all Now I''m from South-Africa,here we have adsl router/modems You set the router to do the dialup and authentication and the set it as your gateways box''s gateway.Now sometimes the links gets drop and is off for a while.Are there any way,for linux,my gateway of letting me now that the link was/is down.Note that the box is not dialing so there is no adsl-status. What I