search for: tryme

Hello, I felt annoyed enough when having to redirect user back to their previous location in a hackish way that I wrote this plugin. It avoids storing POST and Ajax request. It also has a facility to specify actions not to store in the history. If you are interested, it''s there: http://blog.cosinux.org/pages/rails-history See you all, Damien -- Damien MERENNE

...Great job" } evt_tool(1001) { puts "I warned you" } Those IDs: 1000, 1001 may clash with other toolbars on the same frame. For a fixed app this may be not a great problem, but for generated apps it''s a mess. I really want to be able to do tb = Wx::Toolbar.new(frame) tryme = tb.add_tool(Wx::ID_ANY, "Click me", fancy_bitmap) dangerous = tb.add_tool(Wx::ID_ANY, "Don''t click here", fancy_bitmap) evt_tool(tryme) { puts "Great job" } evt_tool(dangerous) { puts "I warned you" } (Well, those ID_ANY should also go away...

Does anyone on the list have a script for notifying pagers that they would be willing to share? I have found a reference in the archive to such a script, but previous attempts to find the author of that posting have failed. Anyhow, I am looking to set up a system whereby a message is sent to a pager when a voicemail is left in a specified mailbox. (This is easy, it's built-in to

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

After having looked around on google, voip-info, etc, I am coming here for a bit of insight from anyone who might be willing to provide some.... I am using * (it's A@H, but this should apply to any * install) to handle my home phone line. I have a TDM11B card, and my Verizon line comes in on the FXO port and my cordless phone is plugged in to the FXS port. Everything works just

I have looked at the wiki and the list archives and I haven't found anything specifically addressing this problem, so here goes: We are using asterisk voicemail to page users (on-call personnel) when a message is received. However, we only have numeric pagers, not alpha-numeric. If I send an e-mail message that has no subject line and has the number I want to appear in the numeric pager

According to https://issues.asterisk.org/view.php?id=16339 , the default value for the dialdebounce parameter of the wctdm module has been changed to 32 and is now user configurable. I have two questions: 1.) Am I correct in presuming that, if the default of 32 does not work for me, I would specify this option in the file /etc/modprobe.d/dahdi.conf (at least for my distro, which is Elastix

Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. 2.) faxdetect=incoming will, upon detection of a CNG tone,

OK, after my last message about fax detection, I feel a bit better informed and able to press forward. I started looking into this because I was getting lots of false positive fax detection errors in the logs with faxdetect=both set in chan_dahdi.conf. Anyhow, I do not currently use fax detection, and we have a dedicated Fax DID on our PRI, so setting faxdetect=no works fine. Having said that, I

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,