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.
