The meat of the issue is SIP phones at remote clients stop ringing, can not call eachother, and sometimes ring/answer for the wrong extension. They can always call out and registration never seems to be lost. It almost always occurs after very heavy call volumes but sometimes before any calls are placed. Rebooting the phones always corrects the problem, as does restarting asterisk or standard re-registration (after timeout).
- Phones: Polycom (430 or 601)
- rtpkeepalive=10 (seems to make no difference if 0, 10, or 100)
- Codec: g729
- Registration: every 3600s on username and secret (works fine)
- No firewalls
- Different internet connections
- Different Routers (but all SOHO)
- Server: Asterisk 1.2.17 with public IP (no nat) in colo (data center)
Some things I have tried:
- Took five phones home and tried recreating issue through Linksys, D-Link, Belkin, and Trendnet. Could not get the D-link to recreate issue so replaced a few clients routers with a d-link, did not correct problem.
- Tried different settings for rtpkeepalive, did not fix problem
- Called Digium (will no longer support open source version)
- Tried limiting to two phones at remote location, did not fix problem
At first it seemed like a network problem (NAT, slow router) but several clients now have this issue and all are connected to this gateway. Clients not connected to this gateway have never reported this issue.
Food for thought
- I suspect that reducing the registration time could correct the issue but this gateway will carry so many phones the load may become excessive
- during compiling zaptel I got the following notice but have found no reference to it anywhere: /usr/src/zaptel-1.2.16/xpp/xpp_zap.c:411:2: warning: #warning “HZ != 1000. PCM would be good only with Astribank sync”
Any help would be greatly appreciated, I am running out of ideas short of higher end networking equipment (not an option). We are willing to hire a freelancer if you wish to offer your services.