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Asterisk Support | Re: Music on Hold - Live Streaming 27 Aug 2013 | 05:02 pm
The main difficult with what you are trying to do is getting the performing rights clearance to use hte material, rather than the technical aspects. Statistics : Posted by david55 • on Tue Aug 27, 20...
Asterisk Support | Re: SIP TLS Handshake failed 27 Aug 2013 | 05:00 pm
Binding both TLS and non-TLS to the same port will cause problems. However, I think bindport only applies to UDP, and TLS is always TCP. Statistics : Posted by david55 • on Tue Aug 27, 2013 5:00 am •...
Asterisk Support | Re: Variable for incoming dialed number 27 Aug 2013 | 04:37 pm
Generally it is always the extensions number. I think you will find that ${DNIS} is coming from your GUI. Having said that, there may be some cases where ${CALLERID(dnid)} is useful. Statistics : Pos...
Asterisk Support | d_channel up and down. console warning 27 Aug 2013 | 04:35 pm
Hello support: I'm receiving strange warnings in my asterisk console: pri_find_dchan: Span 1: D-channel is down! Span 1: D-channel is up! the call are dropping after 10-30 second. need real help... ...
AsteriskNOW Support | Re: sip outbound "busy" howto troubleshoot wanted 27 Aug 2013 | 03:55 pm
I join you. The same problem. Incoming calls are OK. Outbound fails. Peer is up, registration is OK. Different dial patterns were tried, but still "all circuits are busy now, please try to call again...
AsteriskNOW Support | Re: zyxel USG 50 firewall and AsteriskNOW 27 Aug 2013 | 03:18 pm
navaismo wrote: Your local extent shouldn't be NATed and how are the settings for the external extension related to Nat? Navaismo, I'm sorry but I don't understand your question. By "local extent" ...
Asterisk Biz & Jobs | need to change calling mode 27 Aug 2013 | 02:43 pm
Hello, need to change calling mode in asterisk 1.8 as follow: Code: user1 call user2 if user2 is online then normal call else { user1 receive calling tone; some code // ex. sending pu...
Asterisk General | Re: Help addressing single dial plan for multiple servers 27 Aug 2013 | 02:38 pm
vbcrlfuser wrote: What I'm asking for is help with how others may have dealt with the following in one dialplan. Currently SiteA and SiteB each have sip termination and receive 50/50 split of call tra...
Asterisk Biz & Jobs | Selling powerfull softswitch category 4/5 27 Aug 2013 | 01:20 pm
Hello, I am selling with 200 Euro powerfull softswitch category 4/5 routing and billing. Capacity : 500 channels Contain all addons Recording Calls The billing is on mysql you can modify what do you ...
AsteriskNOW Support | googletts not working 27 Aug 2013 | 12:00 pm
this post also exists in the asterisk support forum viewtopic.php?f=1&t=83362&start=0 but it seems to be a problem only in asteriskNOW so I thought I would repost here I've followed the instruction...
