[asterisk-users] SIP resigtrations and OpenSer

[Thomas Kenyon]

I have an ITSP provider that will only deliver calls using SIP 
registrations (would prefer delivery to static IAX or SIP url, but hey), 
periodically their servers don't respond to a renew request, and when 
this happens the sip stack in asterisk (1.4.0) stops working until 
either a SIP reload is issued (or sometimes a restart now).

I'm wondering if this can be solved by installing OpenSER, and using 
that to register with the remote provider and redirect to asterisk 
through a single static sip trunk.

Are there any other solutions that I haven't thought of.

TIA for any help with this matter.