Call Drops after 32 secs

[Robert Birch]

I have a GXP-2135 that I am connecting to a 4.2 FusionPBX server. Both are on the same network.

Internal calls and incoming calls work fine. When I make a call to an outside number, the call will disconnect after 32 secs.

I have a bunch of Cisco SPA-514Gs that are all connected the same way and work fine. I want to replace the Ciscos with Grandstreams.

I have tried multiple settings (No NAT, Stun Server, Random ports, etc) but can't get it to wok longer than 32 secs.

I'm sure I'm missing something real easy.

Thanks for the help,
Robert Birch

 

[smn]

It's almost certainly a NAT issue.

I would go through the NAT configuration settings.
https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal#NATTraversal-FreeSWITCHBehindNAT

I believe you have to set $${external_sip_ip} and $${external_rtp_ip} to whatever your WAN IP address is.

 

[Robert Birch]

I was thinking it was NAT, but both the phone and the server are on the same local network. Therefore I didn't think NAT came into play?

I did try with NAT set to NO, set to Keep-Alive and tried with a STUN server. Nothing made a difference.

Thanks,
Bob

 

[smn]

I was thinking it was NAT, but both the phone and the server are on the same local network. Therefore I didn't think NAT came into play?

I did try with NAT set to NO, set to Keep-Alive and tried with a STUN server. Nothing made a difference.

Thanks,
Bob

If internal calls between extensions work but external calls (which go through NAT) do not then it is most likely NAT. Why are you trying to set NAT to no?? Did you read the link I posted? Did you try the suggestions I posted?

Here is another link. This one talks about one more config in vars.
<X-PRE-PROCESS cmd="set" data="bind_server_ip=my.public.ip"/>

https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal

If you do the things in these links it should work. While troubleshooting, instead of using stun try putting in your public IP manually.

 

[Robert Birch]

I do have $${external_sip_ip} and $${external_rtp_ip} set correctly to the WAN IP. The Cisco 514Gs in the office work fine. It is just the Grandsteam that does not.

I'll try setting the STUN to the WAN IP and see if that makes a difference.

 

[smn]

I do have $${external_sip_ip} and $${external_rtp_ip} set correctly to the WAN IP. The Cisco 514Gs in the office work fine. It is just the Grandsteam that does not.

I'll try setting the STUN to the WAN IP and see if that makes a difference.

I missed the part about it only being the Grandstream. If your Grandstream is not registered or losing registration to the server I believe that will also cause that problem. There may be a session timer setting on the phone. If enabled try disable or visa versa.

If you are watching SIP packets verify the phone is sending an ACK response to Freeswitch at regular intervals. Otherwise I believe session timer will not be reset on server side. Maybe you have STUN enabled on the phone and it's trying to send ACK to the external IP instead of internal.

 

[Robert Birch]

Looks like the phone doesn't send the ACK back to the Fusion box. Tried the STUN setting as the outside IP of the fusion box, no work either. I've got the Grandstream tech looking at it, but still unable to get it working as of yet.

 

[Robert Birch]

Spent some time with the Grandstrem tech and are still unable to determine what the issue is.

I built a test system and was able to make it work quite easily, so there must be something with the original Fusion system causing the issue.

Thanks for the help on this

 

[rusguns]

I used to have that issue but with simple change the settings to TCP in the telephone side I fixed it immediately... you should have a try. You have nothing to loss.