I have only seen this problem when using SIPgate SIP trunks which actually "register". If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of asterisk I've tried. Is there a work around that doesn't involve putting an opensips server between the asterisk server and the sip trunk?
In article <4FECCD0C.firstname.lastname@example.org>,
> On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
> > We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
> > Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
> > and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our
> > PRI to the PSTN and we hope will allow us to failover to other Asterisk
> > servers (ie, Voip2 and Voip3). Voip2 is our current production server,
> > and Voip3…
I have a bunch of different customers on an Asterisk Box (the PBX).
This Asterisk Box is behind another Asterisk box that provides a PSTN
Up to this point I've been using IAX between the 2 Asterisk boxes, but
I would like to use SIP instead.
After doing some testing I have the following issue. If customer A calls customer B, but the call goes out through the PSTN
and comes back in, the call is rejected at the PBX because it wants
I can guess that this…
I am attempting to get an asterisk server to step out of the media
path, but am running into a brick wall. Can someone assist? Here's my
setup.. Ultimate SIP Provider ---> LCR Trunk (Asterisk 1.6) ----> PBX (Asterisk 1.8). I am attempting to get the trunk to step out of the media stream.
There is no NAT involved, all machines have a public IP. In the trunk's sip.conf I have: directmedia=yes
directrtpsetup=yes And on the connection to the pbx I have canreinvite=yes On the pbx I have the trunk connection set to canreinvite=yes. In the…
Recently a have a little problem with a Cisco device, SPA3102. I use
this device with asterisk to dial out with outbound trunk. (SPA3102
has 1 FXO port)
It working ok , but the device SPA3102 do this : when a call is placed
for outgoing in asterisk and send to SPA3102 , this device "answer
and dial the number in the same time" , in my CLI I see the channel
is open , but on the phone I hear the ringing from the provider PTSN ,
then the answer....…