How to configure a SIP trunk between OXE R8 and Asterisk?

[christophe.rdz]

I'd like to test a SIP trunk between one OXE (R8 with sip licences)and Asterisk.Where can I find a documentation about the SIP configuration on both the alcatel omnipcx and the asterisk server to do that?

Thank's.
Christophe.

[cavagnaro]

I'm blind I'm blind I swear I didn't read this post....1 2 3 4 5....calm down deep breath.....

[torrentula]

Maybe I can help...Search first....then post....Don't forget to check the big flashing banner at the top right of your screen. Not much about asterisk there but there is a whole section dedicated to sip on the OXE, imagine that. Then if you know anything about open source, Asterisk has it's own forum dedicated to questions just like yours about it as well.

A basic search with both sip and asterisk in the search on this forum will bring back 8 pages of results. Please start there otherwise Cav might sneak into your house at night while your sleeping and...

[cebollica]

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 created as trunk to use first.
In the extensions, you need to put in the field "direct DID" the same number of the extension (for example if you have created the 301, put in the field "direct DID" 301. This assign a virtual ddi, and you don,t need to create some "inbound routes", only needed if you want to call ring groups, etc.
In general parameters allow "incoming anonymous call" and set the country you need.
In OXE side, of course you need to create the sip trunk group, configure the sip proxy and sip gateway. To have to check if there are 3 sipmotor process (very important), if not you can kill the sipmotor initial process (kill -9 "number of process"). To check this for example try ps -edf |grep sip and you can see all sipmotor process.
After that you can create the sip external gateway and finally create the routing number to call Asterisk (for example 3 if extensions in Asterisk start with 3).
With this all works fine even if you have some release before 8.0.
Try and enjoy

[cavagnaro]

Maybe I can help...Search first....then post....Don't forget to check the big flashing banner at the top right of your screen. Not much about asterisk there but there is a whole section dedicated to sip on the OXE, imagine that. Then if you know anything about open source, Asterisk has it's own forum dedicated to questions just like yours about it as well.

A basic search with both sip and asterisk in the search on this forum will bring back 8 pages of results. Please start there otherwise Cav might sneak into your house at night while your sleeping and...

I'm not that kind dude

[torrentula]

I posted that months ago CAV

I was trying to help with some of the asinine requests being posted. I think maybe he checked the docs after that

[Antonio Carretta]

How to configure a SIP trunk between OXE R8 and Lotus sametime sut lite 852?