When we call a conferencing service an use the phone to Mute our end (instead of the Conference line's mute feature) the Call will get dropped between 5-6 minutes.
The Provider says that since we stop transmitting all together, the Upstream carriers (yes more than one of them) drop our calls.
On the VoIP Tab of the LAN 1 properties (we are only using LAN1)
I have RTP keepalives Settings as Follows:
scope set to RTP
Timeout 10 Sec
Initial Keepalives Enabled
The only other thing that I can think of is the settings on the Firewall. It is an Cisco ASA 5510.
timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00
timeout sip-provisional-media 0:02:00 uauth 0:05:00 absolute
I can try increasing them to see if that increases the time to drop.
Thanks,
Scott<-
The Provider says that since we stop transmitting all together, the Upstream carriers (yes more than one of them) drop our calls.
On the VoIP Tab of the LAN 1 properties (we are only using LAN1)
I have RTP keepalives Settings as Follows:
scope set to RTP
Timeout 10 Sec
Initial Keepalives Enabled
The only other thing that I can think of is the settings on the Firewall. It is an Cisco ASA 5510.
timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00
timeout sip-provisional-media 0:02:00 uauth 0:05:00 absolute
I can try increasing them to see if that increases the time to drop.
Thanks,
Scott<-