Alcatel Unleashed

Dear all
Don,t use R7.0, with 7.1 the problem is resolved. Alcatel says "in 7.1 are some new changes in SIP protocol". I don,t know what kind of changes, but the ALCATEL-ASTERISK union works fine again.
Thank you

Dear aflores
I have made one eservice request to Alcatel about it.
The answer is : "In R7 in fact there are some changes in SIP (adaptation to any RFC without more details)".
I know among Alcatel PBX all is right, in the other side (Asterisk for example) is where Alcatel says we have to correct the ...

Dear all
I have a lot of OXO-OXE systems joined against Asterisk servers with SIP Trunk.
If I use the last release of OXO (7.0), even the last patch, I have a problem: The Ars table apears "active" in the line against Asterisk server, but I can,t call from OXO to Asterisk, the message "unavailable ...

Dear all
I need this tool for investigation purpose, Anyone can post it in pastebin or send my by mail if its posible?
Thank you very much

Hi guys
In Asterisk side, I recommend to use Trixbox 2.2
You need to configure the sip trunk in Asterisk (just put the Cs Ip Main in the field "host"), after that an outbound route (for example if we want to send the "1" to OXE put 1. in the field prefixes) and define the trunk you have previously ...

Hi all
The problem you have is the rights in the voice mail ports. You must add rights in the menu "suscribers-station list-features-part 2- Join incoming and incoming, Join incoming and outgoing, Join outgoing ang outgoing.
Try and enjoy
Bye

Hi all
I just have installed the new OTUC version 5.0 .543, and 4980 works fine, but with Websoftphone I have a problem:
Websoftphone starts fine, I see a message "Starting events" (I use Spanish, so the message is more or less the last one), and after that the message says that are temporary ...

Hi
Be careful with the codec we use in the calls, use G711, if you use G729 you need aditional license in Asterisk.
Bye

Hi all
Using Trixbox version 2.2 , no need to do nothing special neither asteriks and OXO.
In asteriks, when we create some users, put in the field "CID" the same number of user, to assign it a virtual DDI with the user number.
In general parameters in asteriks, allow "incoming annonymous sip", if ...

Hi all
I have all works fine!
I have used Trixbox in version 2.0.1 (it is not the last one, but is very good for me).
I needed to create a sip provider in asterisk, in Web Admin, with the IP of CoCPU.
After that in the file "extensions.conf", context "inbound from sip-external" I have added one line ...