similar to: xorcom hardware good idea?

Dear Asterisk users, We would like to announce that Druid, Open Source Unified Communications project has just made a major release: Druid 2.0. It is out!It has a ton of new features and a highly improved interface. Asterisk stability has also been greatly improved. For more info http://forums.voiceroute.org/showthread.php?t=837 Some of the key features - Improved Web GUI, faster and smoother -

Hi there, I need help in implementing Asterisk with LDAP. I' ve installed Asterik 1.4 with CentOS 5.2 and I would like to use with it an existing zimbra LPAD. thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081018/5f725eea/attachment.htm

I'm impressed with the call log reporting that Switchvox offers. There's a screenshot here: http://www.switchvox.com/sv?cmd=screenshots&pic=13 Can normal Asterisk do this? I installed AsteriskNow on an old computer, but I couldn't find anything like that in the GUI. I found the CSV file in /var/log/asterisk/cdr-csv, but that's not much help unless there's some other

I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

hi i want to ask for your opinion what is better for a call center 100 current calls and other 200 current calls make the server step by step or use a auto install cd like asterisk now, druid elastix ....? and why? Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks

hi i am using asterisk 1.4.22 ubuntu 8.4 i have two Ethernet one for ssh and other one only for voip calls when i start a call using originate in the manager or the cli in the voip Ethernet i get something like 4Mbits/sec of traffic only 1 G711 call. if i start the call using a soft phone everything is normal. any idea? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny

hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi wich predicitive dialer are you using and wich one do you recomend? a link to the project/product and a link to a how to will be VERY apreciated. Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi this is mi first email and just for say hello. David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/375a91e5/attachment.htm

hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part --------------

hi there are a lot of virtualization solution out there and every one "is the best" and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your

hi i need many cheaps atas or some very cheap way to connect analogs phones to asterisk what do you recomend? i searches and only find solutions like 40 U$D (in the states, here in argentina is like 80 U$D) per phone any links or something? thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination.

hi i am integrating asterisk whit an open source CRM or maybe i am integrating an open source crm whit asterisk any way the CRM is vtiger, (www.vtiger.com) i need an opensource queue log parser to put inside the crm agent statics. i need any parser i can port it to php or to java if you don't know any parser maybe you can send me a link or a pdf whit info on how to parse the log. Thanks David

Hi, I'm looking a queue manager compatible with queues.conf. It should allow me to change agents from one queue to another and change it's priority without poblem. Also it must be web based :). Does anyone know any program? Thank you in advance! Santiago del Castillo

I am looking for someone that could share their code for this function: Outgoing call -> macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones -> if human connect to agent else hangup and write status to cdr. Need help with this! Regards / Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL:

I would be grateful if someone could tell me where to find the docs to get started on the following problem: A program needs to be written to place a SIP call to a certain extension on another Asterisk system, and see whether the call state ratchets up to "ringing", then drop, and take action on the results. Can anyone refer me to the appropriate starting point to read up on this?

Hi, I've noticed that if I have a multi-line linksys (942 or 962) phone with the same sip registration mapped to each line key, that if all the lines are full the phone will accept another call. I would expect the phone to respond with "busy" so the call would to directly to voicemail. Has anyone else experienced this and know of a workaround? I know it seems like an

Hi, I just installed R on a new windows 7 machine and am having a probelm with the default libraries. The default libraries are not what I want them to be so when i say install.packages("XXX") the packages don't install where I want them to. Ideally everything would install to the same location as the base packages. When I look at my library paths I get the following. >