similar to: Skype for Asterisk. Any return of experience ?

this may be an easy question, but i am unable to solve it: i want to take the selected choice and process it in the controller: TEMPLATE: <%=start_form_tag :action => ''get_article'' %> <select name=pricelist> <%= options_from_collection_for_select (@customers, "id", "name") %> </select> <%= submit_tag "Get" %>

Hi, I have been studying the R manual on lists but cannot seem to create a simple R list in C - I keep on getting "'dimnames' applied to non-array" whenever I attempt to assign names to the list elements: Wanted output a list structure something like [ type="Bid", price=2.0, volume=1000 ] I can get up to a list of [ "Bid2, 2.0, 1000 ] But for the life of

The aggregate function for some reason will now work for me. The error I'm getting is: "Error in sort.list(y) : 'x' must be atomic for 'sort.list' Have you called 'sort' on a list?" agPriceList=aggregate(PriceList$Size, list(PriceList$bandNum),sum) *Price list dataframe:* dput(PriceList) structure(list(Price = c(0, 8.18, 8.27, 10.42, 10.5, 10.6, 11.13,

>> >> >> Hi All, >> >> After getting licences for Skype for asterisk a while ago I finally got >> around to setting up a server with two channels and setting up a bcp on >> the skype end. >> >> The idea behind this is the following: >> >> Users can dial into the PBX, get authenticated and only after> >>

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

By the way, I am currently testing this product from Skype. I would like to be able to receive calls ona Skype name on our pbx. 1) It works beautifully and you don't have to do anything in particular. 2) It's disproportionally expensive which is why I want Skype for Asterisk to work. SfS costs $5 per month per channel just to test the beta! I find that insane, but I wanted to test it.

Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/fce6307c/attachment.htm

Is there anything special that has to be done to make video calls work? It doesn't seem to work for me (no video). What CODECS are supported?

Hi all I have asked this question before but have not got any helping input. I'm really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to

if I have.... QQQQ<-priceIts("QQQQ",quote="Close") QQQQ<-priceIts("QQQQ",quote="Close");plot(QQQQ) and then i want to do the same thing but say with IBM instead of QQQQ is there an easy way like replace qqqq/ibm Thanks in advance./Jonathan

I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium?s web store and purchase (for zero dollars) the

Hi, In this http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375dating from 2008, experiences with Grandstream GXW4024 were asked. Has anyone something up-to-date to share about this ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091005/ad69fa22/attachment.htm

Dear useRs, I would like to query an Access database through R based on a date attribute but, unfortunately, I fail to do so. For example, the table test_table of the test.mdb looks like: ID cd market competitor dd price 1 1 2007-11-20 atl-bos delta 2007-11-20 210 2 2 2007-11-20 atl-bos delta 2007-11-21 190 3 3 2007-11-20 atl-bos delta 2007-11-22

Hi All, After getting licences for Skype for asterisk a while ago I finally got around to setting up a server with two channels and setting up a bcp on the skype end. The idea behind this is the following: Users can dial into the PBX, get authenticated and only after authentication get access to internal PBX extensions. I CAN do this with a PIN, no sweat, but from a user perspective it

On this weeks VUC call we will collectively be our own guests. That is, we'd like to know what was the big issue that impacted YOU in 2010? All opinions welcome. Here are a few things to get you thinking in advance: - Apple's Antenna-gate - Asterisk 1.8 Launches - Amazon EC2 as a DOS platform - Cisco launched UMI video conference device - More HDVoice capable phones - Skype Outage - VoIP

It seems that Access needs that you surround the dates with a # symbol. You probably need something like. res <- sqlQuery(channel, "select * from test_table where market = 'atl-bos' and competitor = 'delta' and dd = #2007-11-20#") Hope this helps, Stefano -----Messaggio originale----- Da: r-help-bounces at r-project.org [mailto:r-help-bounces at r-project.org]Per

Is there an issue with running it with the latest from the 1.6.2 branch? I did an svn update and make install and now when somebody comes in via Skype, I get an infinite loop of: [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]:

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Hi, We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that "Trapezoidal VoIP is Evil". Join us and ask questions, make comments, argue about geeky details... and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an additional handset. Those of us who have these phones like them a lot. All dial in info is here: http://VUC.me -