To enable the compilation for the addon that is coming with Asterisk 1.8 when doing compilation for the Asterisk, what should I do?
All, I have a sip trunk up and running with a CUCM Express, passing calls fine except for a comfort noise error Im getting on Asterisk: NOTICE: rtp.c:788 in process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn ..
Have not been able to confirm this, anyone heard this from another source? http://bit.ly/jCj5uj We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Sk..
On 05/24/2011 01:07 PM, firstname.lastname@example.org wrote: > I have tried faxing to the DID from 2 different fax machines connected to different POTS lines.One fax machine is a Xerox Workcentre, and the other is a Brother Intellifax.Can you provide some more informat..
I know I asked this some time back, and I got no response then, and neither did someone who asked at the start of 2009 either by the looks of it (other than a reply from me to use a PCI card!!!) However I now have a client whod bought one of these bo..
On May 20, 2011, at 3:30 PM, Anthony Messina wrote: > On 05/20/2011 01:20 PM, email@example.com wrote: >> #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38.Sending faxes is not a requirement.Does anyone h..
Anyone know if its possible to force asterisk to use my_table for
passwords instead of the secret table?
I am looking for tutorial to generate a callfile so that after my program executes, a callfile is generated and pass to asterisk to send to the recipient. Any suggestion? Besides, do you know if there is a web-based GUI to send sms via asterisk? Than..
all, Just in case if anyone will be interested in BroadTel UPA-1, a USB to FXS adapter embedded with SIP softphone. Product specification is as follows: Hardware: USB to RJ11 FXS adapter 1 USB port, for computer connection 1 RJ-11 FXS, for phone connect..
With asterisk 1.4 intermittently I have an issue when i call manually using a hard phone Snom(sip extension) a number X I found another client with another number Y I don’t know if I have any things wrong in my configuration .i have this issue 1/..