restamp
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- Apr 24, 2016
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This is the problem I ran into. The crux of it is that TelecomsXChange sends the INVITE request to port 5060. which falls under the purview of pjsip, not sip. (At least it did on my system: Incredible 13-12.3.) Thus pjsip needs to be told about this IP address.Unfortunately, I cannot get the numbers working. Below are the Asterisk logs. Unnecessary to say that fail2ban is blocking the ip during the call.
Trunk configuration made according to Nerd’s article. Did I miss something?
2018-01-31 07:38:41] NOTICE[16495]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '"1347-mycallerid" <sip:1347-mycallerid@-my serve rip>' failed for '216.66.23.179:5060' (callid: [email protected]:5060) - No matching endpoint found
[2018-01-31 07:38:41] NOTICE[16495]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '"1347-mycallerid" <sip:1347-mycallerid@-serverip>' failed for '216.66.23.179:5060' (callid: [email protected]:5060) - Failed to authenticate
Go to Connectivity -> Trunks -> Add SIP (chan_pjsip) Trunk
Enter a Trunk Name, the SIP server (216.66.23.179), SIP Server Port (5060), and Context. Make sure "Permanent Auth Rejection" is checked so that pjsip doesn't retry Authenticates after the first failure. I also set Qualify Frequency to a large value (32000) to minimize the error messages in the log because that was broken on my system, and for good measure I set Maximum Channels (outbound) to zero.
That's all I needed to do there. After establishing an Incoming Route for each DID, they all worked for me.
Good luck!
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