similar to: OT: does Sipura SPA 3000 support UK caller id?

Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad

At the risk of being flamed.... Has anyone had any success get the 'El cheapo' Wildcard W100P clone's (?20 flavour) to work with UK Caller ID? I'm not sure what the status of Asterisk 1.6 is with respect to UK caller ID, being that we have an odd method of sending the FSK ahead of the ring, but I'm guessing I can't be the first to ask this? Keeping in mind that cost is

Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also seems to not notice any of the line state changes on the PSTN when the remote party terminates the call. It only recognises the offhook signal which gets sent much later. Chris

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS

Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Chris

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

Hi all, Anyone able to remotely reboot their password protected Sipura SPA-3000 from command line. I am trying: Sipura SPA-3000 from command line: # wget http://admin:mypassword@192.168.1.55/admin/reboot The strange thing is it works fine when I go to http://admin:mypassword@192.168.1.55/admin/reboot with my web browser... Thanks....

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

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, I was wondering whether anybody here would help me clarify this minor issue please. If I have the following setup; Asterisk ------ Sipura SPA 3000 (fxo) --------- Pstn Line Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does

All- I'm attempting to set up a Sipura SPA-3000 in Sao Paolo, Brazil. Not being a portuguese speaker, I'm having a rough time of finding the relevant information on how to make the thing pick up the PSTN line and make an outbound call. The sipura in question works fine on a bench connected to a POTS line in the US, but is now plugged in in Sao Paolo. The immediate thing I notice is that

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA

I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much because it's lined up right, but other times you'll hear a really long ring (starts sounding normal, then sounds "weird" -- like two rings played at