similar to: Can I connect device on FXS of Sipura 3000 to internet virtually ? - it can only call ISPs numbers on POTS line

Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to

hi, I have some problem with musiconhold or playtones (background,...) in this context someone dial out thru sipura 3000: Executing Dial("Zap/1-1", "SIP/sipura3000/054419949|20|m") in new stack -- Called sipura3000/054419949 -- Started music on hold, class 'default', on Zap/1-1 -- SIP/sipura3000-61fe is ringing -- SIP/sipura3000-61fe answered Zap/1-1

Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in

I'm trying to get canreinvite=yes to work. I would like asterisk to release the line and let the 2 ports on the sipura device to talk to each other directly. Is there a setting I need to activate on the sipura device, or is there something else I need to do? It's possible that it is a nat problem as the sip device is behind a firewall, but it works fine otherwise. Any suggestions?

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

Hi, I've found approximate same pricing for both. Sipura 2100 seems to have more features... What are differences between those two ? What about their reliability (specially regarding fact, that they deal with analog phones) ? Thanks in advance, regards, Rob.

So far I have gathered that on my NAT (where asterisk server is) I have to port forward UDP ports: 5060 and range 10000 - 20000 to my asterisk server But I'm still stuck how to configure Sipura (behind NAT) what sip proxy and out-bound sip proxy, under which tab to change. -- #Joseph

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 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

HI,ALL, I have multiple PSTN lines registered as multiple SIP channels (e.g. SIP/line1, SIP/line2, SIP/line3, etc...), on the multiple gateways( I uses the SIPURA3000). I wants to arrange them into an ordered hunting group for outbound calling. I used http://www.voip-info.org/wiki/view/hunt-dial+macro for reference. My configure files like blew.

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?

Hello all, Normally I just use pri's with our asterisk systems, but a request came in to add some normal pots lines to the setup. We have 3 lines, and they run into the fxs ports. They hit the dialplan just fine, and they always dial the "s" extension. However, my question would be... Is there a way to determine what number was dialed and have it forward to a specific phone? With a

Hello list, I am currently doing a job for a summer camp. They would like to have several phones around the camp from which people can call in to the main office. It is an older campus and it is comprised of mostly old nungalow type housing. I need to install these phones several hundred yards from where the Asterisk server will be. The way thier telecom wires are currently set up is that they

I'm trying to get an Elographics serial touchscreen working with CentOS 6.2 using the xorg-x11-drv-elographics driver. Touching the screen will wake it up if the screen has blanked, but it never moves the cursor on the screen. There is also a keyboard and mouse connected while I'm trying to get this going and they are loaded automatically and using the evdev driver. Here are the relevant

Hi, at the university department we have quite old Panasonic 2+6 PBX (1BRI + 6 POTS for outputs) and 25 local analog extensions. We would like to add Asterisk with 1 fresh BRI line and possibly integrate with existing equipment (we would like to crossover between both pbxses). What would be most efficient way to do this ? Thanks in advance, regards, Rob.

I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? -- #Joseph GPG KeyID: ED0E1FB7

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

Does anyone have a Sipura 3k running, and using the FXO? I've got things working right, but if I try to toss a *67 in the dialplan, it seems the sipura is throwing a 403 forbidden back. For example: exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/${EXTEN:1} works fine exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/*67${EXTEN:1} does not (even if I toss a couple Ws in) I can't

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....