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.email@example.com>,
> 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....…
I have to make an asterisk gateway in front of several other asterisk.
This gateway will essentialy be used for outbound call.
This gateway will be connected to other asterisk by IAX trunk, outbound
call will use SIP trunk (voip provider or patton isdn).
I have a TE220BF available than i can use for dahdi timing source. Is a
good idea, or this will give me zero benefit for timerfd timing source
(will host this gateway on debian squeeze or centos 6.2) ? Thanks.
Anyone have an H.323 trunk tied between their Avaya and Asterisk box that
works? I am having some issues trying to get the two systems to connect. I
am using the ooh323 channel to try to make the connection between the two
system. I have all my configs if anyone would like to look over them. If I
do a trace on Avaya I get a denial event 1191: Network Failure. Thanks!
I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (220.127.116.11) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager).
The connection seem to be there. When I do a "sip show peers" on the Asterisk server I see the Callmanager as Monitored and online however I can't get any calls to pass from the CM to the Asterisk. If I debug the SIP I get a regular "SIP/2.0 400 Bad Request - 'Malformed/Missing URL'" which is…
On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour
> Hi all,
> I've tried search this problem on the list... no luck...
> The case is:
> without externip/localnet config on sip.conf [general] my SIP trunk works,
> but with no audio NAT problem (asterisk sends the private 192 address to the
> when I configure externip/localnet correctly my SIP trunk simply disappear!
> Checking the signalling with tcpdump shows me that Im…