Hello!I migrated asterisk 11 to 13 as user of FreePBX 18.104.22.168.As customer of German Telekom, I have three numbers and therefore three trunks – each number is bound to one trunk.After migration, some callers complained about missing ringback tone:t..
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 aster..
In article <4FECCD0C.firstname.lastname@example.org>, James Sharpwrote: > 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 (..
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 connection. Up to this point Ive been using IAX between the 2 Asterisk boxes, but I would like to use SIP inste..
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? Heres my setup.. Ultimate SIP Provider —> LCR Trunk(Asterisk 1.6) —-> PBX (Asterisk 1.8). I am attempting to get the tr..
Hy all, 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 aster..
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 isd..
I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6
but I cannt make it work, can any body help me plz?
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 conf..
I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (22.214.171.124) 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 the..