similar to: [OT] spa3000 (Regional & Line1) NL settings required

Hello all. The set up: - Kernel 2.4.18-3 - HP Office Jet K80. - USB port - hpoj-0.8 driver I've been struggling installing and configuring the new driver for Linux/USB printing, due to a couple of silly mistakes. The question here is are there any differences in setting up a print server using this new driver than the usual

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

Hello all. I'm using a HP Office Jet K80 on a RedHat 7.3, kernel 2.4.18-3, using the USB port and Samba 2.2.3a. I've installed the new driver to support usb printing, namely HPOJ-0.80. I can use lpr <filenames with.ps> and it prints fine, also a test page from printconf-tui is no problem. So the Linux side seems to be fine. When I try set up the printer on the windows machine, I

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

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

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

Hi all, Can anybody explain what these values exactly mean. As you all know its the dialtone value on an SPA3000 of linksys. 350@-19,440@-19;10(*/0/1+2). Can anybody help me how to write this code for a dialtone of frequency 425 which is continous. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Hey, all... If this is too off-topic, I'd be grateful for directions to a more appropriate mailing list. I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000 which is registering twice with Asterisk - once for its FXS/Line1/VoIP1 and once for its FXO/PSTN/VoIP2. My eventual goal is to have inbound calls on its FXO ring four times on its FXS and then fail over to

Hi, I have a little situation here :( Perhaps somebody can give me a hand with it. I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor

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

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

Arrggghh. Tearing my hair out here. I'm trying to set up the spa3000 in the UK for my home, and want * to control the dial plan I've googled to no avail. I've read the manual to no avail. Can someone, please let me know what the parameters is the spa and * are to a) receive a call from the pstn b) make a call to the pstn from the phone attached I can make sip to sip calls (i.e. I

Hi, I have a little situation here :( I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor my router have DynamicDNS. So, if I

Hi, I'm using a SPA 3000 as FXO and FXS termination connected to my * box. I'm using the caller ID prefix trick explained here: http://www.voip-info.org/wiki-Sipura+3000. All seems to work really fine, there is only a problem when I hangup my IP phone after a conversation and the other party don't: in this case the PSTN line remains connected until the other party hangs up. Any

My porblem is incoing PSTN calls are being forwarde to the * box, the phone rings, but when the phone is picked up, the call is not taken - it continues to ring. I am forwarding the call to (<S0:6666>) in my dial plan Can anyone assist? This is driving my crazy! Extract from the * console Executing Dial("SIP/3001-047c", "SIP/2004") in new stack -- Called 2004 --

Hi, I'm using a Sipura SPA 3000 to receive calls from a PSTN line and to send them to Asterisk. But on Asterisk, I don't see the PSTN caller ID! I only have User ID of the SPA 3000 as caller number. The caller number is present on the PSTN line. I'm in france, maybe the SPA 3000 is not compatible? If you have an idea, please tell me! Thanks, Micka?l

Please forgive is this has been posted, but without a good search function the archives are hard to search. Have a home network - two laptops XP PRO, one Linux RH 7.3 and one NT box. All in the workgroup named workgroup. I would like to rid myself of the NT box as all it does is spool a printer and a USB Hard drive for backup purposes. Network hooked to a Linksys Cabel Modem router, So NAT

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