Changing DNS responses

Hi,

when using Telekom Company Flex it is possible that the IP-Adress changes.

I have the problem that it looks like that FreePBX is using different IP-Adresses for Registration and calls when the IP adress returned by the dns query has changed. Then i receive an 403 Forbidden because my FreePBX is not registered at this server.

[2022-11-07 08:42:54] VERBOSE[39741] res_pjsip_logger.c: <--- Transmitting SIP request (920 bytes) to TLS:217.0.139.239:5061 --->
REGISTER sip:tel.t-online.de SIP/2.0

[2022-11-07 08:42:54] VERBOSE[2595] res_pjsip_logger.c: <--- Received SIP response (842 bytes) from TLS:217.0.139.239:5061 --->
SIP/2.0 200 OK

[2022-11-07 08:44:06] VERBOSE[15905] res_pjsip_logger.c: <--- Transmitting SIP request (1097 bytes) to TLS:217.0.150.112:5061 --->
INVITE sip:+49<CENSORED>@tel.t-online.de;user=phone SIP/2.0

[2022-11-07 08:44:06] VERBOSE[2595] res_pjsip_logger.c: <--- Received SIP response (354 bytes) from TLS:217.0.150.112:5061 --->
SIP/2.0 100 Trying

[2022-11-07 08:44:06] VERBOSE[2595] res_pjsip_logger.c: <--- Received SIP response (461 bytes) from TLS:217.0.150.112:5061 --->
SIP/2.0 403 Forbidden

Does anybody has a similiar problem and/or knows how to solve this?

Greetings

Jens

Any one with a ‘dynamic’ ip does, you will need a ‘dynamic dns’ service and a timely method to check for when it changes so to update Asterisk

I’d note that, if your diagnosis is correct, the provider is going beyond the requirements of SIP. SIP only requires registration for incoming calls. However, I believe quite a few do check against the registration before accepting outgoing calls.

Also, could you confirm that dicko has correctly interpreted this a being an unstable source address from you, rather than a provider with multiple incoming servers who don’t share information between themselves.

ISPs who change IP addresses other than after a significant period of disconnection are always going to be a problem for SIP users, and the best option is to find a business oriented ISP who will, ideally, provide static, address, or, at least, will renew with the same address.

Also, could you confirm that dicko has correctly interpreted this a being an unstable source address from you, rather than a provider with multiple incoming servers who don’t share information between themselves.

No, i have a static ipv4 address. It is definitly the ISP’s IP that is changing (or FreePBX selects a lower priority server because another one is not reachable):


ISPs who change IP addresses other than after a significant period of disconnection are always going to be a problem for SIP users, and the best option is to find a business oriented ISP who will, ideally, provide static, address, or, at least, will renew with the same address.

I understand, but Telekom is the largest ISP in Germany for private and business.

In their technical guidelines they are considering the problem of changing server IPs (paragraph 8.1.2): https://www.telekom.de/hilfe/downloads/1tr119.pdf

In my opinion FreePBX (or asterisk) should always use the server where it is already registered or renew the registration on the new ip before calling out.

Here is someone with a similiar problem: Re: [asterisk-dev] Enforce Registrar Stickiness for operation on (3GPP) IMS networks

Such behavior goes against the general specification for locating SIP servers, you’re not supposed to do such a thing so that load balancing can occur if the DNS arrangement of the provider allows it along with failover. Functionality COULD be added by someone, but it would need to be behind an option.

That specific provider likes to do things their own way generally against specifications.