SIP Phone problems ( Register but no Calls)

Good day.

i have a client that bought a SIP phone. and has a subscription for the sip. trough his isp.

we turned off the sip-alg . the phone registers but does not get calls in or out.

was suggested to do dst-nat to the phone on 5060 and 5061 ports to the phone. then some other installer said that its dangers to forward ports to phone. we must use a stun server?

please assist.

Your client must configure Outbound Routes and Inbound Routes from his PBX if his phone registered if I’m not wrong

he uses his ISP that gives him the sip service

Ask for support directly from the service provider?


Personal note, do not read

I don't understand why the "technicians, friends, and relatives" who come to tinker at my clients' homes,
instead of asking me directly for the solution to the problem (after all, they PAY me to provide the service, assistance IS owed...),
start acting like experts and instead of "lowering themselves" to ask me directly for help,
they start looking for some tutorial on YouTube, asking on forums, or asking Artificial Deficiency...

we spoke to the supplier of the SIP Line. and all they say is it is on our equipment. i did the setup on the mikrotik. with 2 split networks. and the everything works on both networks. except the new sip . we installed the sip app on client’s phone and tested it and that works out of the box.

Make sure you reboot after turning off SIP-ALG

Calls themselves use two protocols,
SIP(tcp or udp depending on your setup 5060 5061 )
and
RTP(udp only typically 10000–50000 or 1024–65535)

If SIP is working correctly the phone will ring. If calls either drop, audio issues, audio one way issues (then its a RTP issue)
If Phone does not ring at all you have a SIP issue and that would depend on your setup (ie NAT,firewall rules etc etc) where the sip protocol is not going through.

Try SIP on either TCP or UDP on the phone to see if the results differ or not.

Test with Microsip to eliminate the physical phone being an issue.

Current versions of Ros have both sip and rtsp helpers.

I assume rstp defaults to turned off.

You could try with both turned on, and perhaps all 3 of the other possible options.
sip on, rtsp on
sip on, rtsp off
sip off, rtsp on
sip off, rtsp off

Both off would be the common case.

Some providers may expect you to have port forwards on your router.
Though often work pretty well without them.