similar to: Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?

Hi, I have an older phone with touch screen from Philips. It have ti 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 and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come

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

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?

Hi, I'm an absolute newbie and wanted to know the following. I want to have a setup where I have a PSTN line connected to my Asterisk box and want to know if it is possible to make more than one simultaneous outbound call through that VoIP gateway? Can Asterisk do this magic of concurrent calls on one PSTN line?? If I put it in other words then can I receive more than one simultaneous call

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

Anyone have any ideas as to why a call coming in won't ring the phone? I can call the phone from my cell and when I hear it ringing on the cell phone I pick up the house phone that should be ringing and am able to talk. I have tried two different pap2-na adapters, have verified the ports on my firewall and also a couple of different house phones. I am not running Asterisk yet but will be

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.

I would like to use a genuine Windows 2000 server as my PPTP server (getting poptop to work correctly is annoying the hell out of me) which needs to authenticate off of a Samba server configured as a PDC. A Windows-based PDC has a flag for "dialin access allowed" for each user. I searched the Samba archives and see several posts over the years that say it "sounds easy enough to

Hi, after some corrections in my settings IAX2 dialin seems to work now. I get the incoming call, but i cannot here anything or can speak. (If I take the call the other side see that the connection is established if I close the call the other site is seeing it too) If I press hold in Idefisk the other side can hear MoH but not me. Asterisk print in the CLI interface that he starts MoH. The

I have a samba ldap pdc set up. (2.2.8a). I have a windows domain member that is joined to the domain running ras. When users try to dial in to the server they get "Does not have Dial in permission/rights". Is this an option that samba recognizes at this point? Is there a way to tell windows that a user has dial in privileges? Sean Cook Kinex Networking Solutions

On a SIP phone is it possible to enter the dialplan when the user picks up the phone without having to wait for the user to press an extension? Is is possible to do something like [sip-test] s,1,Answer s,2,Playback(welcome) s,3,WaitExten(30) 1,1,Noop(exten 1) ... t,1,Goto[s,2] --------------------------------- Be a PS3 game guru. Get your game face on with the latest PS3 news and

Hi, i was able to solve my problem. During my playing around with * and capi i changed several options in config files. I did this while my * was running. To test if my changes where successful i entered "reload" on * console. This didn't help. But after i stopped asterisk and startet it again, everything worked perfect. So it seems that doing a reload while asterisk is running

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

Hi All, There are two option in QoS settings of the SIPURA ATA. ( I can't just remember them). please tell me what is better and which one should I choose for my DSL line (128kbps) with a small LAN. thank you kumara

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

Hi all .... I Have installed Bering LRP on Many sites and I am very pleased with the capabilites of shorewall. Howerver I came across a prob that I am unaware ot its solution. Using shorewall 2.0.2f Kernel 2.4.24 On one Site LRP box serves internet outgoing connections through ( static IP ) a DSL line AND an incoming dial-in PPP conection. My shorewall configuration Is based upon the fact

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout

Hello, I am trying to use a HFC-PCI (CCD/Billion/Asuscom 2BD0) card in TE mode to dial-in and out with ISDN. The problem is I can not get the card to dial out with a Zap channel. Dial-in is working. I am using bri-stuff 0.1.0-RC4 (but tried also RC3 and RC2k). I tried all combination of "immediate", "overlapdial", "pridialplan". I earlier also managed to dial out

Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten =>