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Hi All I have 3 GrandStream BudgeTone-100's which connect to an * box with a HFC-S based ISDN card using ISDN4Linux. I have setup the BudgeTone-100's to use Early Dial which for calling between the three phones works well, but for the external calls using the following extension exten => _9.,1,Dial(Modem/g1:${EXTEN:1}) Only the first digit is dial on the ISDN Line. Does anyone know of

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb

From what I understand, shared interrupts are permitted on the PCI bus. Whether they work for a specific card or not depends entirely on how the driver is written. Someone at Digium should know this. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106

...he 192.168 net). It goes without saying that the asterisk box must also have a public IP address so that it can synchronize itself with a remote time server. In my setup, I have one net card for the public address, while the 2 192.168 addresses are on a second card. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106

...sn't matter whether I reboot from the phone's web site or using the menu reset function: either works. I have only tested this with a private NTP server. This may explain why some of you have not been able to get anything but the default NTP server to operate. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106

I know the recommendation it to not run more than 2 T/E100P's in a system but what about X100P's.. Usually there are 5 PCI slots in a system, has anyone tried 5 x X100P's in a system? Later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

I've just finished a general purpose configuration utility for the GS phones: 1) Generates files from scratch (using MAC), from HTML config listing, or by directly downloading from the phone. 2) Does multiple simulteneous edits. 3) Can reboot as many or as few phones at a time as you like. I would like to offer it to the list, but there are 2 issues: 1) I want to GPL it first, if

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but