Hello,
As I see this setup aren’t up to date for the new 1.8 and 2.8 from PBX, and where I’m so newbie, no idea how setup this…
Any help please to update this for asterisk 1.8 and FreePBX 2.8?
I’ll be really thanks,
I know this was about 5 years ago but I just had the same issue when making outbound calls via Skype Connect trunk stating “all circuits are busy now plseas try again later.”
What I did in the outbound dial pattern was use international dial. For example instead of 9|NXXNXXXXXX use 9|1NXXNXXXXXX.
That solved my problem.
Haven’t M$ blocked all third party access to Skype now? If not I’d be interested in a good and reliable way of doing this as one of my customers was asking about this the other day.
At this moment I have one of my trunks in FreePBX registered to one of my Skype Connect’s SIP Profile and it is making and receiving calls just fine.
Thanks. I’ll give that a go.
Here is the config that comes up if you enter “asterisk trunk” as the search term at community.skype.com, you may have more success with this. I also have to ask where did you come up with nat=auto, can you point me to the example that shows that? If it’s on this web site it needs to be taken out. There is no auto NAT option.
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=
fromuser=
secret=
disallow=all
allow=ulaw
allow=alaw
nat=no
fromdomain=sip.skype.com
insecure=port,invite