Incoming calls No matching endpoint found

I use trunk wich take only ip for use, no name, no password.
This is my trunk

type=friend
host=XXX.XX.XX.XX
port=5060
context=from-trunk
insecure=port,invite

this i get then i call to my number, that i take from provider

[2018-12-13 13:26:01] NOTICE[23079]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '"XXXX902967" <[sip:[email protected]](mailto:sip%[email protected]);user=phone;cpc=ordinary>' failed for '[11.149.6.36:5060](http://11.149.6.36:5060/)' (callid: [[email protected]](mailto:[email protected])) - No matching endpoint found

<--- Transmitting SIP response (572 bytes) to UDP:[11.149.6.36:5060](http://11.149.6.36:5060/) --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 11.149.6.36:5060;rport=5060;received=11.149.6.36;branch=z9hG4bK4hf9rp207gpihct2ikg0.1

Call-ID: [[email protected]](mailto:[email protected])

From: "XXXX2902967" <[sip:[email protected]](mailto:sip%[email protected]);user=phone;cpc=ordinary>;tag=v7j0n2qydh

To: "3310062" <[sip:[email protected]](mailto:sip%[email protected]);user=phone>;tag=z9hG4bK4hf9rp207gpihct2ikg0.1

CSeq: 488 INVITE

WWW-Authenticate: Digest  realm="asterisk",nonce="1544707561/f66a5f22ff15bd53cde752763c260e2a",opaque="2b021079082ac873",algorithm=md5,qop="auth"

Server: FPBX-14.0.5.2(13.19.1)

Content-Length:  0

<--- Received SIP request (414 bytes) from UDP:[83.149.6.36:5060](http://83.149.6.36:5060/) --->

ACK sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 11.149.6.36:5060;branch=z9hG4bK4hf9rp207gpihct2ikg0.1

CSeq: 488 ACK

Call-ID: [[email protected]](mailto:[email protected])

From: "XXXX2902967" <sip:[email protected]:5060;user=phone;cpc=ordinary>;tag=v7j0n2qydh

To: "3310062" <sip:[email protected]:5060;user=phone>;tag=z9hG4bK4hf9rp207gpihct2ikg0.1

Max-Forwards: 68

Content-Length: 0

You’ve created a chan_sip trunk but the incoming invite is going to the pjsip port. Config your provider to send calls to the correct port.

i create pjsip trunk and incoming calls work.
Now out calls have no sound.

Sounds like your signaling is resolved, but RTP ports are not connecting. Are the RTP packets being blocked?

One-way audio is almost always a problem with NAT. Make sure all of the NAT settings for all of the devices in the network are set correctly, and make sure that the firewall/router is forwarding all UDP traffic from 10000:20000 to the PBX from the external address.

i have problem with regestration on provider.
Provider say what i must have no registration,
i can calls using only ip addres
If exist any analog in pjsip trunk?
chan_sip option insecure=no

I use pjsip now, and it create problem.
[2018-12-13 15:22:38] WARNING[23079]: res_pjsip_outbound_registration.c:792 schedule_retry: Temporal response ‘404’ received from ‘sip:XXX.X49.6.36:5060’ on registration attempt to ‘sip:[email protected]:5060’, retrying in ‘67’
How using interface can i solved it.
I should not use registration gequests.


Sip server i delete for secure) Sip server=ip of my provider

when i reseve call (from telephony to out)

  == Spawn extension (from-pstn, XXXX2902967, 1) exited non-zero on 'PJSIP/Megafon2-000000b3'
    -- PJSIP/Megafon2-000000b3 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
    -- Called PJSIP/[email protected]
    -- PJSIP/Megafon2-000000b3 is making progress passing it to PJSIP/112-000000b2
       > 0x7faf6cabc360 -- Strict RTP learning after remote address set to: 10.69.0.91:53028
       > 0x7faf644948e0 -- Strict RTP learning after remote address set to: 83.149.6.36:10936
    -- PJSIP/Megafon2-000000b3 is making progress passing it to PJSIP/112-000000b2
    -- PJSIP/Megafon2-000000b3 is ringing
    -- PJSIP/Megafon2-000000b3 answered PJSIP/112-000000b2
    -- Channel PJSIP/Megafon2-000000b3 joined 'simple_bridge' basic-bridge <cf9b6ad0-54a3-4a5c-a827-614cf0df3edc>
    -- Channel PJSIP/112-000000b2 joined 'simple_bridge' basic-bridge <cf9b6ad0-54a3-4a5c-a827-614cf0df3edc>
    -- Channel PJSIP/112-000000b2 left 'simple_bridge' basic-bridge <cf9b6ad0-54a3-4a5c-a827-614cf0df3edc>
  == Spawn extension (macro-dialout-trunk, s, 31) exited non-zero on 'PJSIP/112-000000b2' in macro 'dialout-trunk'
    -- Channel PJSIP/Megafon2-000000b3 left 'simple_bridge' basic-bridge <cf9b6ad0-54a3-4a5c-a827-614cf0df3edc>
  == Spawn extension (from-internal, XXXX2902967, 6) exited non-zero on 'PJSIP/112-000000b2'

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