how to connect asteriks and OXO with SIP Ip trunk

[cebollica]

Hello
I want to connect asteriks and OXO with SIP trunk, but i Have some problems. I can call from OXO to asteriks user,s (like softphone sip X-lite or Wifi linksys phone), but at the moment i hang up the call is rejected. The same if i call from asteriks to OXO phones, the phone rings and i see the user in display, but the call is rejected at the moment i hang up.
I have tried to change all the parameters I know in OXO and asteriks (i have a lot of experience in Alcatel VoIP), but I can,t do any more.
I should need help
Thank you very much

[alex]

There are not many parameters to change in OXO
Check SIP trace in *

Code: Select all

asterisk*CLI>sip set debug

Also how OXO is declared in sip.conf?

[cebollica]

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 with this argument: exten => _.,1, DIAL(200@from-internal-custom,20,1). With this, all calls from OXO to asteriks goes to sip user 200 (XLITE softphone for example).
In OXO the parameters in ARS are the same to another OXO call.
See you later

[cebollica]

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 not, the call from OXO is rejected.
Don,t forget to create a Trunk or provider in asteriks with the CoCpu IP as destination of calls, and to add an "outbound route" with the prefixes we want to dial to OXO througt asteriks sip trunk.
Enjoy
Bye

[henk1964]

Works from OXO R 6.1 to *
From * to OXO still have 503 service unavailable...

Any hints?
Trixbox latest version

Thanks

[cebollica]

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

[henk1964]

Got this one working!

Had to set the asterisk phone range to private and now it works both ways.
Protocol was already at G711, problem was in the private setting.
Hope this helps anyone.

Thank you all for your help!

[iverona]

Hi henk1964!

I'm trying to acomplish the same setup as you did (connecting our OXO to Asterisk through SIP trunks). The problem is that I've no experience with OXO (but lot's of experience with Asterisk) and our telco provider has no experience with Asterisk.

Which hardware setup do you have? I think we have a OXO Rev2, so maybe the first thing should be upgrading to Rev5 o Rev6. Which cards/modules are you using for the VoIP? My telco partner suggested us to buy the following:

3EH73063AB VOIP16-1 CARD
3EH73013AB PLACA HIJA SLANXA
3EH73048AK COCPU-2 1.00

What do you think? What do we need? Should we better try to link using PRI (QSIG) instead of SIP?

Thank you very much in advance. Regards,
Ignacio.

[henk1964]

Start with an upgrade, the release 2.x does only support H323.
You can not upgrade to R5, only latest version upgrade is possible.
From R2.x will realy cost you, ask your local Alcatel partner.

Henk

[iverona]

Thanks for your reply henk. Finally we are going to try H323 as the "cheap&quick" option, so lets see if it works.

Thanks again!