[Asterisk-Users] ISDN-Phone (HFC) <=>*<=>SIP-Provider: audio only in one direction, no nat problem

[Uwe Betz]

Hi List!

I have an interesting problem. I am behind a NAT Firewall which works 
fine with SIP. I am connected to T-DSL in Germany and there the 
DSL-Connection is interrupted every 24hours and buck a few seconds later 
with a new dynamic IP.
My Asterisk is registered with several SIP-Providers and this works 
fine. In addition the *-Server has a HFC-S ISDN interface card installed 
in NT mode (using zaphfc) with an ISDN-Phone connected.
Everything works fine when I make a call from the ISDN-Phone through my 
SIP-Provider to other Phones or SIP-Users (external).

BUT: As soon as I get the new IP (either automatically due to the forced 
interrupt of my DSL line each 24hrs, or manually forced) every still 
seems to work fine but audio goes only in one direction from now on 
(alwasy I can't hear the other party but they can hear me and signalling 
also works fine). So I make a call, but while the phone I am calling 
rings I can't hear the ringtone in my phone. When the other side answers 
the phone I can't hear them while they can hear me loud and clear.
A "relaod" on the CLI solves the problem till next IP-Change.

I know there were already some things reported with dynamic IP's but in 
most cases nothing worked anymore after the IP changed. What can I do 
(maybe with the settings ind some conf-files). In addition I found that 
if I have srvlookup=yes in my sip.conf Asterisk can't register with my 
sip provider. But many example configs dsay you should use srvlookup=yes 
and I hoped that this might solve my problem but I can't use this 
setting set to yes at all.

Any ideas on what to try?

Thanks,

Jui