similar to: ISDN and DVO

I am trying to set up a linux based faxing solution for a client, and have found that the modem they have (ancient dataplex external unit) just isn't up to the job. It talks to some remote fax machines but not others. A new external modem ranges from AUD$75 to AUD$400, which got me thinking of other possibilities... #1 FXO PCI card (more expensive for 1 port, probably cheaper for 2+) #2

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

More an echo algorithm question than a purely asterisk one... I have the following setup: Handset - PAP2 - Asterisk - SPA3000 - Telco And no matter what I do, I get echo on a call routed out via the PSTN when I talk into the handset, in the order of a hundred ms (my estimate, could be wildly inaccurate!). Echo will occur also when I have a handset plugged into the phone port on the SPA3000

http://www.zoomtel.com/products/voip_products.html anyone using these? they look very interesting in that they support ilbc, and they offer a separate cheaper model without g729 license. i'm wondering if their EC is better than the spa3000, and what the echo tail length is. -Dan

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

I'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret=1111

I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4. Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo of spa3000, all works ok. Then I call from a sip phone configured for using g729, to the fxo of spa3000, it also works ok. The problem is that after this, when, making again a new call from spa2100 to spa3000, spa2100 receives only white noise. I suspect a

I was incorrect in a previous email... The situation in question is this: Asterisk <---BRI---> PBX <---BRI---> PSTN There are Samsung extensions on the PBX and SIP extensions on Asterisk. I want to be able to use TAPI to initiate dialling, and the PBX has no such feature so Asterisk must initiate it. For an Asterisk initiated call from a PBX extension to a PSTN number, this works as

Hello, I'm looking for an ISDN BRI-U interface for use in the US. I'm primarily interested in using the BRI as a trunking interface into the PSTN with Asterisk. Naturally, cheaper is better. I currently use a Nortel Norstar system with BRI-U trunks, and really like the digital PSTN interface. Would really like to replace the whole mess with Asterisk but want to keep the ISDN trunks.

http://bugs.freedesktop.org/show_bug.cgi?id=11906 Summary: X server does not start with nouveau driver Product: xorg Version: git Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org

Hello. I have been trying to duplicate these u32 matching rules using TCNG, but without much success: tc filter add dev eth0 protocol ip parent 1:0 prio 1 u32\ match ip protocol 6 0xff \ match u8 0x05 0x0f at 0 \ match u16 0x0000 0xffc0 at 2 \ match u8 0x10 0xff at 33 \ flowid 1:10 Wondershaper 1.1a implements this and if I run it things seem to be shaped as expected. But this TCNG

Hi, all A quick question. I have SPA3000 and trying to get call waiting to work. I do receive call waiting tone, however hook flash does not seem to work. I think, I set up SPA3000 correctly. Basically, doing HF 2 (switching calls in Australia) does not do anything. Is there any examples on how to setup hook flash operation in Asterisk? Thanks, Rudolf

http://bugs.freedesktop.org/show_bug.cgi?id=12582 Summary: Crash on GeForce 7300 SE Product: xorg Version: git Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: major Priority: high Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy: panov

Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Hi, I am thinking of starting a project at Work to pilot to use of Astrisk and VOIP, and would like to tie in several projects together. Currently we are looking at purchasing an ISDN and H.323 based video conferencing system from Polycom. It suggests the use of 3-4 BRI lines (ie 6-8 B channels, but in the form of individual BRIs). We need ISDN for the video conferencing to communicate with

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Is anyone in the world making gsm 'picocells' which could be connected to an Asterisk server and allow gsm mobiles to roam to them (and therefore become just another extension) when in the office? Obviously lots of things to consider (it's a licensed band) which I think was the big holdup last time I asked this question anywhere. I know there was talk about using them on aircraft

Voxilla.com has a great config wizard for the SPA-3000 and * http://voxilla.com/spa3kasterisk.php I took the output from this wizard and dumped it on my test box with an SPA 3000 (with some mods to match my * contexts) and everything worked great. Calls from the PSTN to the spa3000 are routed to dialplan #8 on the spa3000, which dials * Both the FXO and FXS port register with * The SPA3000 is

Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab -> Dial Plans -> Dial Plan 8 (<S0:66610>) Should I put extension [66610] in sip.conf with a context in extensions.conf that