Calls dropping recently

We’ve been using Asterisk for many years, and since 2012 we’ve been using Flowroute after trying a couple of other VOIP providers. Never had any issues until about a couple of months ago. Around the same time Flowroute discontinued their Nevada PoP and we reconfigured our Asterisk server to point to the US EAST VA PoP. I’m not sure if this is related to the issue though.

What’s been happening recently is that calls just drop randomly, and this occurs more frequently with longer conference calls. Flowroute is telling us this:

“The issue was the rport was present initially in the invite and then on the 200 OK it was no longer present. Because of this, we (Flowroute) changed our signalling to the IP in the contact header. If the rport is kept throughout the call then we will continue to signal to the original IP and the call should not end. If you do change the configuration we can look at the call to see if that does fix the issue.”

I’m not sure what configuration needs to change. We haven’t touched the configurations for a very long time now (the only change was made to the host and fromdomain settings below (from sip.conf). As far as I know, the dropped calls issue had started occurring before we even changed these settings.

; ####################################################
; ## Flowroute Settings ##
; ####################################################

[flowroute]
type=friend
secret=xxxxxxxxxx
username=xxxxxxx
host=us-east-va.sip.flowroute.com
dtmfmode=rfc2833
context=inbound
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=us-east-va.sip.flowroute.com

We also tried changing the nat=yes setting to nat=force_rport,comedia but that didn’t help at all. The asterisk server is behind a firewall, and all the phones are in the same office behind the same firewall as well.

[general]
context=default
allowoverlap=no
srvlookup=no
jbenable = yes
jbmaxsize = 200
jbresyncthreshold = 1000
register => xxxxxxxx:xxxxxxxxxxxx@us-east-va.sip.flowroute.com
externip=aa.bb.cc.dd
localnet=192.168.15.60/255.255.255.0
;nat=yes
nat=force_rport,comedia

Hoping someone can provide some assistance.

I would suggest getting a SIP trace using “sip set debug on” of an actual call failure that was dropped, to determine precisely who and why.

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