Bart Coninckx
2010-Jun-17 20:26 UTC
[asterisk-users] calls dropped after 20 seconds in a non NAT situation
Hi all, I have a rather old 1.4 installation that recently was connected to a new network via an IPSEC tunnel. No NAT-ing is involved anywhere (I've seen posts about the same phenomenon but with NAT). It first the phones on the PBX network did not get the audio of the phones on the remote network, but that was fixed by removing an externip entry out of sip.conf. What I have now, is that all calls are cut after 20 seconds. The log files say: Maximum retries exceeded on transmission 629be818-f372199a at 192.168.10.104 for seqno 101 (Critical Response) It seems Asterisk gives up after 20 seconds to wait for some answer from the phone. What do you guys think is the way to proceed here: change the configs somewhere or upgrade Asterisk? Thank you, Bart
Possibly Parallel Threads
- Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?
- Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk
- Sip proxy registration help
- SIP registration timeout
- Asterisk behind NAT and Kamailio --> Internal IP in SDP and not "externip"