Hi Guys,

I am using a free softphone voip provider and after seconds of a call the calls is dropped, I've opened up the following ports that the softphone uses but I still receive 30 second dropped calls, My ata voip adapter has no issues on the router board.

/ip firewall nat add chain=dstnat dst-address=*.*.*.* protocol=tcp dst-port=13000 \
action=dst-nat to-addresses=192.168.88.180 to-ports=13000

/ip firewall nat add chain=dstnat dst-address=*.*.*.* protocol=tcp dst-port=6060 \
action=dst-nat to-addresses=192.168.88.180 to-ports=6060


Thanks for your help.

a few questions

1) why do you use tcp and not udp because most time it is udp for voip
2) why user port 6060 i assume being for sip most of the time is port 5060
3) i hope port 13000 is for rtp but does your softphone configure for symetrical rtp
4) and a last do your provider send sip keep alive ?

a+
thierry