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 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
