My voip services are not working. I have altered the ip address for two trunks yet the system log reports attempts to connect to previous addresses which I cannot find any configuration for in the GUI but have discovered are in the sip_registrations.conf file.
The old addresses are listed here in the Asterisk Info:
Using Chan_SIP with Flowroute is not the right way to do this. Each one of those PoP’s have over 26+ addresses on them. Chan_SIP only uses one of those IPs. In order to receive calls from that PoP you are registered to, you need to allow all the IPs from the PoP. That means you’ll need around 26+ Chan_SIP trunks for a single PoP.
OR You setup a PJSIP trunk and put all those in the Match field and PJSIP will accept calls from all of them.
Bottom line, providers are using subnets and SRV records to handle failover and other routing. Chan_SIP is not a viable option for modern VSP/ITSPs.
Sure, I have it but for awhile the word was to avoid it due to some problem or other. I have just not had a time when I wanted to switch everything over and deal with any changes that might arise.