search for: easycall

...lity for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi support at easycall.gi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101101/9526607d/atta...

...tp://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 "Not acceptable here" back from (PeerIP) -- SIP/easycall-838e is circuit-busy ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060127/5be0ac94/att...

...lity for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi support at easycall.gi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110127/80e38aab/...

...lity for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi support at easycall.gi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101101/47ded5b2/atta...

...lity for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi support at easycall.gi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101101/6479311a/atta...

Hi All, I?m trying to do ?things? after my Dial application terminates (e.g. play IVR to called party, calling party, etc.). I?m trying to use the local channel for this purpose but so far with no success. I?m using 1.6.1.18 and this is my extensions.conf: [Internal] exten => _22,1,Dial(Local/${EXTEN}@CW/n) ; 22 is test number exten => _22,2,Noop(After Hangup) [CW] exten =>

Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk)

Hello List, I have few installations out there based on 1.6.1 or above. I'm trying to play different voice mail messages based on certain criteria's. For example, I want during office hours to play (in short): "we are not available to take your call, please leave a message", during off-hours and weekends I would play: "we are closed, our opening hours xx:xx-yy:yy, please

Hi all, Could someone please tell me what is the relative "cost" in using conf files oppose to the astdb? Basically I need to match a name to a phone number in order to have all users registered by name and not by number (which I understood is not a good practice). I have 2000 users and a complex dial-plan and server resources become an issue. I could implement this via a context in my

Hello All, As one with theoretical knowledge in programing, but never on Linux, I can understand terms and code structure but I don't know: 1. What shell commands (e.g. ./configure, make, make install etc.) should I run to recompile Asterisk (same version)? 2. What shell commands should I run if I want to apply a change to source code? 3. Is there a general guide on how to upgrade Asterisk?

Hi all, Can the Asterisk do ?things? not during a call? For example I would like to change my dial plan during certain hours\dates or I would like to check some information in the astdb (e.g. counters of al sort) and handle it as required and so on. All of this is not call-related therefore I don?t know if I can somehow do it using the dial-plan applications\functions. I know I can do chron jobs

Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is