similar to: Need help on how to programmatically call an extension & test call state

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

Hello Centos user. Im learning about c/c++ , so far every thing works great. The problem begins when I try to install a new library or a third party api. For example now I need to install the jsoncpp implementation in my centos box but I have no idea how to do it. I realized that my knolowge about those things is very poor. I would like to ask you for some documentation about libraries

Hi, i've found you have some trouble with the spanish tranlation in some of the docs. I don't know if im wrong for writting to this list but i'd like to say that i can help with the trasnlations , i'm from Argentina so my mother languaje is spanish. -------------- next part -------------- An HTML attachment was scrubbed... URL:

My grateful thanks to whoever can guide me in implementing this... I have a need to place calls via Asterisk Manager Protocol to a legacy PBX and twiddle its MWI lights. By some means I get notified that an MWI light has to be changed. Via AMP: 1. Send an ORIGINATE command honking up an incoming port on the PBX via a TA804. a. I presume that the TA804 should be a SIP channel. yes/no? b.

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

Hi there. My name is Lucas and I''m from Argentina. Firstly, forgive me for my english since it is not my native languaje. Now, I''ve been reading on how to compile iproute2 and found that I need to add the following to my kernel, which in my case it is 2.4.20, and it is also the one which this document I read talked about: CONFIG_NETLINK=y CONFIG_RTNETLINK=y # CONFIG_NETLINK_DEV

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 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 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 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'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 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 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 i am thinking to use some xorcom hardware i only used digium hard... what do you think? it work ok? what about echo? it is plug and play only reload or i need to restar asterisk ? i understand the drivers are in the trunk of asterisk, is this rigth? have any one a conf file of a xorcom device? any other important comments are welcome. Thanks David -- (\__/) (='.'=)This is

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 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:

I started two weeks ago with rails and im amazed of what can be done in a short time, i already finished the agile development with rails and some of the tutorials at the documentation section of the rubyonrails.com site. But i realize that many of the tutos and examples around the web and books are using earlier versions of rails (0.x.x) and some of that code and coding practices are

I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) --

On libvirt 6.8.0 and qemu 5.1.0, when trying to live migrate "error: internal error: Failed to reserve port" error is received and migration does not succeed: virsh # migrate cartridge qemu+tls://ratchet.lan/system --live --persistent --undefinesource --copy-storage-all --verbose error: internal error: Failed to reserve port 49153 virsh # On target host with debug logs, nothing

On Jan 1, 2014, at 1:52 PM, Ariel Wainer wrote: > Very strange! It seems like the completly powered down and OK status > yeild the same reading. The reading didn't change after plugging back in > the AC. Well, as strange as that is, I don't think it would be a common case. Ordinarily, the load would be the computer monitoring the UPS, and if the load is off... We can certainly