My task is to receive calls from my sip operator with my asterisk server # 1 than to transfer it to asterisk server # 2 and than to my softphone.
Something like this -
people calling my mobile number (which i rent from my sip operator)
Sip operator receives this call and transfers
Asterisk server # 1
Asterisk server # 2
Myself being very happy )))
As for now i have already configurated my asterisk server # 1 to receive calls. I tried to make an extension on this server and connected my softphone to it. I could receive calls with no problem.
After that i tried to connect asterisk server # 1 with asterisk server # 2 via sip trunking. I created 2 trunks (1 trunk on server # 1 and 1 trunk on server #2) with registry string so they registry on each other. Registration seems to be ok because sip show registry shows that they are registered.
After that i thought that its my lucky day and everything would be ok and start to configure inbound routes. On server # 1 i have chosen destination - Trunk (which i use to registry on server # 2) On server # 2 i have chosen my extension which is connected to my sofphone.
I thought that as i need only to receive calls i shouldnt configure outbound routes as i wouldnt make some calls anyway (only receive). So i have made 0 outbound routes.
Of course, as i am newbie nothing works. When i am calling my virtual number i receive a message - Your call can not be completed as dialed, please check the number and dial again.
So, dear experts in Asterisk and Freepbx please give me some advices on next questions -
Does these sheme has any sense? As i understand i need to make 1 trunk with my sip operator and 1 trunk with registry on each of servers. Than i should put trunk (with registry to server # 2) in my inbound route of server # 1. In server # 2 i should only have a trunk to connect with server # 1 and an extension (in inbound routes) to my softphone. Does it have any sense?
To connect with my sip operator i use username 02301 ( which i received from him, as its my sip id which is connected with virtual mobile number). To connect with server # 2 i use another username officesip with another password. Is this correct? Or i should use on server # 1 only 1 username both in trunk with my sip operator and with trunk to connect with my asterisk server # 2
Do i need outbound routes if i need only to receive calls from my virtual mobile number,
I would appreciate any advice. I realize that i am very very newbie and i think that my questions are stupidies you have seen but please be patient and help with info. Thanks. And good day to all of you.