Incoming calls No matching endpoint found

[necto_random (Random)]

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

[lgaetz (Lorne Gaetz)]

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.

[necto_random (Random)]

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

[comtech]

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

[cynjut (Dave Burgess)]

@necto_random

Now out calls have no sound.

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.

[necto_random (Random)]

@cynjut

. 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

[necto_random (Random)]

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.

[necto_random (Random)]

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/XXXX2902967@Megafon2
    -- 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'