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/dav/myemail/events/
user=myemail
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 get the announcement and/or MOH
blanked out after a random period of
Hi,
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 difficulty is that astribanks devices get different usb device
ids during their initialisation process, requiring hot
On Sat, Oct 29, 2011 at 3:14 PM, Eric van der Vlist v...@dyomedea.com wrote:
Hi,
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 difficulty is that astribanks devices
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/dav/myemail/events/
user=myemail
secret=mypassword
refresh=15
Hi 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 channel to
another, which would leave me with only
Try turning on the Sip debug for the PSTN call as well as RTP debug. Paste
the output here.
The Dial server is connected to multiple 4-port Redfone devices for
handling PSTN incoming and outgoing calls. Outgoing calls always
originate from and incoming calls always terminate at the SIP
On Sunday 30 Oct 2011, Sammy Govind wrote:
Try turning on the Sip debug for the PSTN call as well as RTP debug.
Paste the output here.
There's no SIP involved until the call is picked up -- only PSTN and
IAX2. No RTP log available either.
The Dial server is connected to multiple 4-port
