similar to: Source for 50-pin amphenol cables?

Ernest, I hadn't thought of doing that, though having that added protection would be nice. However, what I'm trying to do it have an incoming call at my home number follow me to my cell phone for selected numbers -- Since I already have three way calling, I'd like get Asterisk to essentially three way my cell phone into the call (or my office number, etc.) I understand the

A little off-topic maybe, but it's still for the Adit used with Asterisk. ;) I wonder where I can buy 50 pin Amphenol cables, with connector on one side, and open cables on the other for mounting in our own patch panels. In Europe, or Sweden preferably. It's said to be very common on telcos, but I can't find it for sale anywhere! Thanks!

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

# Get a list of product ids from a form, then # sum the total price product_list = params[:product_ids] total = Product.find(product_list).sum(&:price) I understand the first list line and part of the second line, but what does the & symbol mean? TIA.

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

We're interested in leasing roughly 15 Polycom IP-600 phones. Does anyone have a vendor that they can recommend for this? Scott

Does anyone know a place in Australia that sells a power supply suitable for the IAXy? I haven't had any luck tracking one down. -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

Is there a way to include: _NXXNXXXXXX _NXXXXXX _011. _911 into my current plan:

assert_tag smells bad. Hpricot smells nice (apricotty!). Testing your views with Hpricot: http://www.lukeredpath.co.uk/index.php/2006/07/07/testing-your-rails- views-with-hpricot/ Big thanks to _why for a great little library! Feedback is appreciated. rSpec integration coming soon (hopefully!). Cheers, Luke Redpath contact@lukeredpath.co.uk -------------- next part -------------- An HTML

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

Are there deltas for the revisions of the PDF version of the Agile book? -- fxn

Hello, So all the pieces are finally here, and I'm ready to play. I remember reading on this list that the connection Channel Bank <-> T100P requires a "reverse cable." Is this a regular Ethernet reverse cable (i.e., only a couple of pairs reversed?) Please help me before I blow something up! Saludos, Jose.

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

Hello all, Just starting to setup asterisk v 1.4.11 and need to run three distinct phone systems for three different companies. So far, I have inbound lines going to the appropriate dial plan within the extensions.conf file. I'm using exten => _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10}) to determine which number is being dialed by the caller and then using a gotoif to get to

How do I solve the problem with between patterns: _1800 _1NXX I would like all numbers 1800, 1877 etc to go through iaxtel but all other numbers 1xxx via voipjet Example in my extension.conf I have: [iaxtel] exten => _1700NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten => _1888NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten =>

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These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

I''ve installed rails (and it works) and now I want to be able to connect to a mysql database. The database.yml file tells me to install the mysql drivers by typing: gem install mysql This is the error I receive: Could not create Makefile due to some reason, probably lack of necessary libraries and/or headers. Check the mkmf.log file for more details. You may need configuration

I''ve got something that I''m not sure is actually doable. I have a very complex model relationship with a lot of parent/child/grandchild/ greatgrandchild etc stuff going on. Is there a way to do a route like this? tld.com/ projects/:project_name/:sequence_acronym/:shot_number/:department/:eleme nt_name/:version/ where: project is the parent of sequence sequence is the