Ive had this same issue.. while im not running on astlinux on my production
sites.. ive found that I needed to do one of 2 things..
anchor the media at the asterisk server, so the directmedia = no on the SIP
trunk, and in the SIP settings I needed to have my externip = my.public.ip.here
and localnet = 172.16.0.5/255.255.252.0 (use your localnet).. it seems that
the forward sends the Internal IP of asterisk in the redirect for the forward..
I Ihandle my forwarding a little different. I handle it on the server as
opposed to the phone.. (I have XML apps that forward the call in asterisk and
light the lamp on the phone when the user presses the key).. those sites
where the call is forwarded at asterisk vs the phone itself seem to fare a
little better.. the reason I do this is the call still forwards even if ther
phone gets unplugged, and the user can use our web-based User interface to turn
their forward on / off remotely or change it.
in one case. the SIP provider is velocity, I had to actually Answer(),
establish audio then transfer the call before 2 way audio would establish.. I
accomplished this by answering then playing a quick "please hold" then sending
the call off to the external number..
I can give you nore details if you wish to do this method-Christopher
On Monday, March 30, 2020, 4:09:09 AM EDT, Tom Chadwin
<[email protected]> wrote:
Morning all
We have a SIP trunk, with ample channels. With staff at home, they have tried
to forward their phones to their home numbers. They do this on the endpoints
themselves (Snom D305). However, it's not working. An incoming external call
hits the phone, and is forwarded, rings, and is answered. However, after
pickup, there is no audio either way.
Can anyone think of any reason why this might be happening?
Thanks
Tom
Tom Chadwin, ICT Manager
Telephone: 01434 611530 Mob:
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