TIPS legit calls are being rejected

[Aaron Outhier]

Getting the following in the logs:

[2021-02-01 07:55:24] VERBOSE[21174][C-00000001] pbx.c: Executing [REDACTED@from-sip-external:1] GotoIf("PJSIP/anonymous-00000000", "0?baddomain") in new stack
[2021-02-01 07:55:24] VERBOSE[21174][C-00000001] pbx.c: Executing [s@from-sip-external:6] Log("PJSIP/anonymous-00000000", "WARNING,"Rejecting unknown SIP connection from "") in new stack

Among other things. The domain name of my PBX is set correctly in my Trunk Provider settings. My provider's FQDN is whitelisted in IncrediblePBX with "/root/add-fqdn Flow-WA us-west-wa.sip.flowroute.com" and I have verified that this is the fqdn from which the calls are originating with the service provider. I am getting zero incoming calls, and my customers are complaining. What's going on? This has apparently been going on for months!

 

[Aaron Outhier]

My callers are receiving the zapateller message.

 

[billsimon]

Do you have a PJSIP trunk configured for this provider? If so can you share the settings?

 

[Aaron Outhier]

Modified from the original FlowrouteLA

type=friend
qualify=yes
insecure=port,invite
host=us-west-wa.sip.flowroute.com
fromdomain=us-west-wa.sip.flowroute.com
dtmfmode=rfc2833
disallow=all
directmedia=no
context=from-pstn-e164-us
canreinvite=no
allow=ulaw

 

[Aaron Outhier]

Trunk is configured for IP auth, so no registration is used. PBX is on a cloud server with a fixed IP.

Incoming calls are directed to the FQDN of the PBX box.

 

[Aaron Outhier]

I just noticed. The following is listed under [chan-sip] under Asterisk Info report:

Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
flowrouteOR               34.210.91.112                               No         No             5060     OK (27 ms)
flowrouteWA               147.75.60.160                               No         No             5060     OK (28 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

However, the actual calls mention PJSIP. Would this make any difference?

 

[billsimon]

Yes, it means the incoming calls are coming to the port where PJSIP is listening. You can do one of several things to make it work:

1. Figure out which port chan_sip is listening on and instruct Flowroute to send the calls there. The incoming calls should then match the chan_sip trunks you have set up and get routed as expected. I am guessing that PJSIP is listening on 5060 and chan_sip might be listening on 5160.

2. Set up a PJSIP trunk to match the incoming calls, using the same details for which you set up the chan_sip trunk. (no registration or authentication)

3. You already have Allow SIP Guest enabled (which enables the PJSIP/anonymous endpoint), so if you then also enabled Allow Anonymous Inbound SIP Calls, these calls would get routed through your dialplan. Not really recommended; it's a catch-all solution and would potentially route any traffic that hits the PJSIP port of your PBX according to your Inbound Routes.

For the longer term, consider standardizing on either chan_sip or PJSIP, with PJSIP being my recommendation because it's "the way forward" with Asterisk. chan_sip is no longer in active development and is considered deprecated in the latest version of Asterisk.

 

[Aaron Outhier]

Whaaat? I never noticed that trunks were either chan-sip of pjsip. I thought Trunks were just Trunks. I see now that all of the pre-defined ones are chan-sip.

Well, I tried creating a pjsip trunk for flowroute, but got lost - too many questions, not enough answers.

I also tried tacking :5061 to the end of the fqdn for my pbx box in my trunk settings, but they said the fqdn was invalid. They don't appear to support non-standard ports for sip traffic. For the time being, I've set everything back to chan-sip. No sooner than I did, I started getting junk calls! Great!!

I recently picked up a number that used to be mine a few years ago. I think I remember now why I ditched it!!