We have a VOIP environment, please see the attached image. We use Asterix 1.6 VOIP server running on Debian Linux, 2 pcs of Juniper SSG-5 routers with 6.3.0r5 firmware, and a Linksys VOIP phones. We have been installed and configured the Asterix server, configured trust-untrust policies on the SSG-5 routers, and set up the phones. All phones in SITE1 have been registered on the Asterix server located in SITE2 via IPSEC tunnel.
Our problem is very strange, i try to explain it:
1. When phone turns on, registers well, incoming and outgoing calls works (rings and voice).
2. After approx 5 minutes, incoming calls will be fail, because the phone rings, but no voice when picks up, and the IP phone displays still "Answering ..." and the caller phone shows "Calling ...". After we take down the IP phone, the caller shows still "Calling ...", until call expires because time-out.
3. If I call back the caller from the IP phone, the call works (rings and voice as well).
4. After the calling back the incoming calls work well, but after approx 5 minutes idle time still no voice of incoming calls, until I make a call from the IP phone to outside.
The phone has been registered continuously.
We checked the follows:
- turned off SIP feature in the ALG in the SSG-5;
- defined voice protocol group in SSG-5 (UDP traffic between port 10000:20000);
- created trust-to-untrust policy in each SSG-5 which allows ANY traffic between the IP phone and Asterix server in both side and vice-versa;
- turned off any iptables firewall in the Asterix Server (all packets accepted);
- logged traffic in SSG-5 by policy, and ensured no dropped packets.
Any idea or suggestions please?