Douglas, You;re right, that method is useful only for one-to-one call but as soon as the call gets transferred etc etc as you mentioned everything will get mixed and confusing. Any way I this can be done? Can’t a call be passed off from one chan..
On Sat, Oct 29, 2011 at 3:14 PM, Eric van der Vlistwrote: > > > Xorcom astribanks get initialized straight on when using Ubuntu 11.10 > packages but I am having a hard time to get the same result running in a > qemu/libvirt image. > > The first difficu..
On Saturday 29 Oct 2011, Raj Mathur (राज माथुर) wrote: > [snip] > Callers coming in from the PSTN (through the Dial server, over IAX2) > can also talk normally after an agent has picked up the call. > However, callers from the PSTN ..
using asterisk-10 on CentOS I am trying to get googleapps calendar integrated with my system. However, following all the instructions that I can find it still fails. this is my config file: [myGoogleCal] type=caldav url=https://www.google.com/calendar/da..
I just had a weird experience.My Asterisk installation stopped working, and upon investigation I found that the ownership of several files had changed from asterisk:asterisk to root:root.The files in question were: /etc/asterisk/extensions.conf /etc/asterisk/features.c..
Samsung shipped 27.8 million smartphones in the last quarter, taking 23.8 percent of the market, Milton Keynes, U.K.- based Strategy Analytics said in an e-mailed statement today. Apple’s 17.1 million shipments, comprising 14.6 percent of the mark..
Sammy, thanks for the response. So based on your recommendation, does this mean that all log lines relating to a given call will retain the same Channel Identifier String for the entire life of the call, even as it moves from the external SIP trunk provid..
I noticed Asterisk 184.108.40.206 execute number dial twice
== Using SIP RTP Co..
For future Google searches: Ended up working with MixMonitor, but I had to remove the b option of MixMonitor (record only when call is bridged). I still haven`t figured out how to make the r option of MeetMefunction properly, but I`ll give up sinc..
Team, i have been facing issues with sangoma card with 16 E1. used LibSS7 asterisk 1.6 with 8 E1 the links are stable , but moment i add another card of 8 E1 for to support 16 E1. link keeps fluctuating any idea why ? Please help Thanks Vinod Dharash..