SIP problem with incoming calls
Posted: Thu Dec 08, 2016 12:07 pm
Hi there,
i am new to Mikrotik routerboard and i don't know much about the configurations. Just the basic stuff, NAT, DHCP etc.
I have problem when transferring calls from an Asterisk server to another in 2 separate location. I have created the NAT for both public IP's of the location and also the SIP Trunk between the two servers. It worked fine for a long time. A week or so now i get this problem, either the call goes to the other location but no voice either the call doesn't arrive at all. The configuration is the same as it was when it was working fine. While searching i came up with some posts that SIP service ports might create problems with incoming calls. So on the MT router that accepts the calls in location B i disabled it and on the MT router that sends calls i leaved it enabled. When this problem occurred i had to restart the MT router on the sending location A. After the restart all it was fine! Until today. It is really frustrating that one configuration that worked fine doesn't work just it did anymore.
I am guessing that the problem is on the MT router on the sending location A... From the Asterisk CLI i get the message "Unable to create channel of type SIP", which make me think that there is some NAT or Firewall Problem...
NAT Configuration
Location A (sending calls): dst-nat of IP of location A with IP of location B with a range 5000-35000 (SIP Service port is Enabled) - MT Routerboard v6.33.3 RB951-2n
Location B (receiving calls): dst-nat of IP of location B with IP of location A with a range of 5000-35000 (SIP Service port is also Enabled) - MT Routerboard v6.19 RB951Ui-2HnD
Any suggestions please? This problem is driving me nuts...
i am new to Mikrotik routerboard and i don't know much about the configurations. Just the basic stuff, NAT, DHCP etc.
I have problem when transferring calls from an Asterisk server to another in 2 separate location. I have created the NAT for both public IP's of the location and also the SIP Trunk between the two servers. It worked fine for a long time. A week or so now i get this problem, either the call goes to the other location but no voice either the call doesn't arrive at all. The configuration is the same as it was when it was working fine. While searching i came up with some posts that SIP service ports might create problems with incoming calls. So on the MT router that accepts the calls in location B i disabled it and on the MT router that sends calls i leaved it enabled. When this problem occurred i had to restart the MT router on the sending location A. After the restart all it was fine! Until today. It is really frustrating that one configuration that worked fine doesn't work just it did anymore.
I am guessing that the problem is on the MT router on the sending location A... From the Asterisk CLI i get the message "Unable to create channel of type SIP", which make me think that there is some NAT or Firewall Problem...
NAT Configuration
Location A (sending calls): dst-nat of IP of location A with IP of location B with a range 5000-35000 (SIP Service port is Enabled) - MT Routerboard v6.33.3 RB951-2n
Location B (receiving calls): dst-nat of IP of location B with IP of location A with a range of 5000-35000 (SIP Service port is also Enabled) - MT Routerboard v6.19 RB951Ui-2HnD
Any suggestions please? This problem is driving me nuts...