Outgoing calls seize to transmit audio in both directions after 50 secons

FreePBX Installation / Upgrade
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PLEASE HELP!!!

Hi, I’ve installed ELASTIX (Free PBX) a while back for a small business, incoming calls are working without any issues, outgoing calls are successfully connected and you are able to hear and talk (person on the other end can hear and talk as well), however, after 50 seconds (ranges from 40-90 sec) audio seizes, neither party can hear nor talk, call will stay connected for over 24 minutes as long as neither party hangs up, once one party hangs up/disconnects the call, the call is ended. My VOIP Service Provider is Vitelity, they’ve been giving me the run around for quite some time, they now are telling me that it’s related to my router, that the ports are possibly not configured properly or “nat=yes” needs to be configured, they say it’s related to the RTP packets being dropped and not transferred properly back to the FREE PBX server.

I double checked my settings on the router and outbound trunk settings, and they are configured accordingly, all incoming and outgoing UDP and TCP 5060-5061, 10000-20000 ports are set to forward to the VOIP (Free PBX) server, the trunk settings is configured as such…

“type=friend
dtmfmode=auto
username=xxxxxxxxxx
secret=xxxxxxxx
fromuser=xxxxxxxx
trustrpid=yes
sendrpid=yes
context=from-trunk ; (this could be ext-did or from-pstn as well)
canreinvite=no
nat=yes
host=outbound.vitelity.net”

Any help would be greatly appreciated!

Thanks in advance…
.

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they are probably correct in that this type of issue is almost always related to router/firewall issues. If your router has any form of SIP ALG or related on it then it is likely he culprit and would need to be turned off.

If you wanted to try a SIPSATION trunk for the FreePBX service we offer, then you could use it with the SIPSTATION module which gives a lot of diagnostic information including what we are seeing on the server side.

In addition, we can change whether or not we proxy the media through the SIP server or direct to the media gateways as well as giving you more detailed information as to what is being seen on the server side if you have issues.

However, given your description, it is probably your firewall deciding that some session has expired and cutting off the RTP traffic give the time intervals you have described.

Good Luck, these can be tricky to solve if you don’t have a view from both sides, Vitelity should be able to tell you if they see any changes on their end after that time.

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