Archives : January-2015
Im investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. Im investigating, but can anyone provide some feedb..
allWE have some users that turns off their phones when they are not at home.We see the warning message: Unable to create channel of type SIP (cause 20 – Subscriber absent)just after the Dial() command and a Everyone is busy/congested at this timemessage.Wh..
HiWere using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues.For a particular customer, when I run queue showI get the following numbers: has 0 calls (max unlimited) in ringall strategy (17s holdtime, 94s talktime), W:0, C:175, A:..
all,Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works.My network setup by ..
Everyone,I am required to write a java program that will get our asterisk to:* Query the database for phone numbers* Loop through numbers and dial* Play message* Get dial pressed response- If 1 = Yes- If 2 = No- If 3 = Connect to Agent* AMD Capab..
while most of the physical phones have keys to handle attended and blind transfer, most soft phones have no support for it. Asterisk offers afeaturemap to assign a key to blindxfer and atxfer and they work fine if the call is still in the same start..
I am trying to setup a WebRTC connection to asterisk 1.13.0. Using Bria a regular SIP connection works, but using sipml5 on chrome, Igot nothing.My network setup by the way: I am working behind a comcast cable modem, the test setup is at digital oce..
does anyone have a recommendation for a SIP phone, which allows dialing from a phonebook, and hiding the dialed number from the end users? Also from the call history of course.It seems Mitel can do this, and I have a use case where this is a requirement.Than..
Im currently evaluating asterisk 13 (Currently on 11).Were testing the migration from SIP to PJSIP.Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect816.749.2815 (Internal: e..
I am trying to setup a WebRTC connection to asterisk 1.13.0. Using Bria a regular SIP connection works, but using sipml5 on chrome, Igot nothing.My network setup by the way: I am working behind a comcast cable modem, the test setup is at digital oce..