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About mini cell network

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psilvao:
Hello, I want to do a small project for a cellular network using 1 yatesbts + 1 BladeRF + 1 Sip Server for incoming and outgoing calls

For this and given that I am new to the use of yatesbts, I would like it in a rural area where there is no other company, to get the cell phones that are on, I understand that using the regular expression .* This is achieved, but as I'm going outgoing call to the SIP server ?, and the same for an incoming call to the cell phone that hooks up with the yatesbts station?

What happens, if I now for the project i use 2 stations yatesbts + 2 BladeRF + 1 Sip Server and the person walks with his cell phone and gets stuck from the first station yatesbts to the second station yatesbts, the call is lost?

Thanks in advance
Pablo

Monica Tepelus:
Yes, the call is lost. The feature that does this is called handover (moving one call from one bts to another) but you would need a YateUCN for this to work.

psilvao:
Thanks Monica, about the other questions, it's possible for example using a SIP Server as Asterisk, Freeswitch for incomming and outgoing calls?


Thanks in advance
Pablo

Monica Tepelus:
I did not understand the question. YateBTS has a SIP server (the one from Yate) already there. If you set Outbound connection then all calls that can't be routed inside (to registered IMSIs) will be routed to this Outbound connection. This is set in accfile.conf in section [outbound].

[outbound]
protocol=sip
username=091
server=192.168.10.1
password=f56543
ip_transport=UDP
enabled=yes



 

psilvao:
Hi again

I'm sorry if I was not clear enough in my query, although YatesBTS has a SIP server which can be enabled or not, in my question it pointed out if another SIP server could be used, such as asterisk, freeswitch to name a few.

If the answer is yes, how do I do the routing or mapping between a subscriber of YatesBTS and the external server, since for YatesBTS the subscriber will have an extension X, which I imagine should be mapped to the external SIP server, in order to generate the call to a PSTN.

If the answer is No !, and the SIP Server of YatesBTS should be used in a mandatory way, how is it controlled that some and not all subscribers can make outgoing calls to a PSTN network?

Thanks in advance
Pablo

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