ok guys, My Asterisk server is up and running....
Here is the steps taken to implement asterisk. My Setup- Virtual Number(DID)---> Asterisk Server---> Intervoip----> Mobile/Landline NOTE:-Intervoip is used only because the calls needed to be routed to our normal PSTN network (i.e Landline/Mobile). If you are comfortable with Softphne(x-lite) or IP Phones you can skill the intervoip settings. 1) Created SIP trunks for incoming connections with only incoming details, leaving the outgoin setting in the SIP trunk as blank. Also the register string was left blank as DID server provider dint allowed SIP registration. 2) Created an Extension " Generic SIP Device" Give an "Display name:" Should be unique Give "OutBound CID" as your virtual Number Secret field is the Password for the extension. You can enable Voice Mail settings if you want, also you can give an email notification if a voice mail is generated.Also give voice mail password in the password filed. 3) Now create an "Inbound Route" for the Extension Just select the extension radio button and the extension number just created in the above step. You can test this with the softphone for incoming connection. Download x-Lite, Right click and got to SIP account settings Enter the Display name, can be anything Username: Your extension you created Password: Same as the "SECRET" field entered in extension Auth User: Your extension you created Domain: IP address of your asterisk server Dial you virtual number, X-lite should ring and you should be able to pick up the call. Till now we configure Virtual DID ----> ASTERISK Now below are the steps from Asterisk to Intervoip and to Mobile/Landline 1) Create a new SIP trunk, but now leaving the incoming setting as blank and filling the details with the out goin setting. Outgoin setting you should get this from your service provider. Enter the register string: A common format is :"username:passw...@domainname/username " But this format may differ from ISP to ISP. 2)Enter the Dial Pattern as per you need. If you are making routing calls to INDIA, mobile phone the dial pattern can be: "91982." without inverted commas. 3) Create an "Outbound route" this time Enter the dial patterns same as you did while create a SIP trunk in the previous step. 4)Select the trunk sequence In our case we will select the trunk we just created before creating Outbound route. Click on submit. Please make sure that you apply all the setting that appear on the top in the red bar. 5) Create a ring group Give a unique number for "Group Description" Enter the number you want to route the call to i.e your fianl destination number 91225453454# Please make sure that you end the number with # Select Ring strategy as "Ring All" 6) Final step, Goto Inbound Route which we created earlier, right in the bottom select ringgroup which we just created. Now if everything goes well, and if you have balance in your intervoip account, when you call the virtual number, the call will be routed on "91225453454" If you need any help on this please feel free. Also a special thanks to Ragu for landing a helping hand...:) Thank and regards Vipin Valsaraj The Quieter u become, the more u will be able to listen...... --- On Tue, 27/4/10, vipin valsaraj <[email protected]> wrote: From: vipin valsaraj <[email protected]> Subject: [LinuxVadaPav] Help Asterisk To: [email protected] Date: Tuesday, 27 April, 2010, 5:11 PM Hello every1, In process of setting up asterisk server, and right now dont know how to go abt it. Set up is... Asterisk server is running on CentOS 5.4 And Asterisk GUI is FreePBX Virtual number--->Asterisk Server-->Intervoip- ->Destination number. 1)Virtual number is a paid service and allows call forwarding, it is directly mapped to the asterisk server from the account settings. 2)In Asterisk server, i have created SIP trunk for the Virtual number with only incoming details filled in leaving blank the outgoin details and Format string since ISP for virtual server doesnot allow SIP registration. 3)I have created inbound route for this SIP trunk 4)Created IAX2 trunk for Intervoip with only outgoin settings and outbound route for IAX2 trunk But when i call the virtual number, its coming on the server but get the reply that the number is not in used. Any help...if i m missing out on something... .. Regards, Vipin Valsaraj The Quieter u become, the more u will be able to listen...... [Non-text portions of this message have been removed] [Non-text portions of this message have been removed]
