Sip Trunk Setup Trix Box Callback

We have got a SIP trunk from over SIP provider and we are using 5 DIDs. Sous Le Soleil Saison 1 Torrent 411. I want each DID to route to a different extension but i don't know how to do it. Our SIP provider told us that are routing the call on user id basis and i should change the dialplan and context to route calls to different extensions.

In your FreePBX, SRV Lookup should be enabled: Do so under the 'Settings' tab, under 'Asterisk SIP Settings' and the tab 'Chan SIP Settings'. Under 'Connectivity' - 'Trunks' add SIP trunk. Specify the name of the trunk and go to the tab sip settings. Enter the following information: 111111: Your sip.

I did all the steps but whenever an incoming call comes in it is routed to only one extension all the time or to none of the extensions. This is what our SIP provider is sending us in header. Via: SIP/2.0/UDP 101.50.;branch=z9hG4bK1636b2ec;rport From: ';tag=as39de4115 To: Contact: Call-ID: 4c4817eb47c786bb70af1e1f6dfcea30@11.20.64.79 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 70 Date: Wed, 20 Feb 2013 10:32:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 213 I have been working on this issue for more than two weeks i have searched so many communities but without any luck. Any help will be greatly appreciated. Imran Newsterisk Posts: 5 Joined: Wed Feb 20, 2013 4:26 am •.

You want one DID trunk, not multiple simple trunks. DID is misused in the Asterisk world. Real direct in dialing means that you get part or all of the dialled number forwarded to you. If you can't get them to provide that, try using a callback extension on your register line, to try and recreate the original dialled number. The ITSP clearly doesn't understand Asterisk, and, in particular, that it can only use the source IP address and port for discriminating between trunks - actually that is a characteristic of SIP, itself. Moves Like Spencer Posts: 12570 Joined: Fri Sep 26, 2008 5:03 am.

I no longer understand what you mean by DID and SIP trunk. Most people misuse DID to mean a number in the direct in dialing range at the ITSP, which is fowarded on a dedicated trunk to the customer, often on the assumption that they are not running a PABX. If you really have only one trunk and it is not a proper direct in dialling one, with different extensions for each of your PSTN numbers, there is probably not enough information for anything to distinguish between the PSTN numbers. Assuming for the moment, you have the more common problem where there are multiple accounts with the same remote IP and port, but on which the dialled number is not being forwarded, the parameters of register are explained in the sample configuration file, and as this is a common problem examples can be found by searching the archives. Moves Like Spencer Posts: 12570 Joined: Fri Sep 26, 2008 5:03 am. Your problem is that your provider is sending you INVITE messages addressing SIP:s@your-ip-address.

If the caller dials 123456, your provider should pass the call to you by sending their INVITE message to SIP:123456@your-ip-address. If they can do that, the 123456 part of that string becomes the extension that is matched against the exten = 123456,1. Line in your extensions.conf file. You then place a line like that for each of the expected extensions. Right now, they're only using extension s for all inbound calls, so you cannot route based on the dialed number. This is not DID! Oldsterisk Posts: 78 Joined: Mon Dec 04, 2006 7:56 pm Location: NJ USA •.