No incoming calls - how to debug

Hi there,

I’ve installed the FreePBX distro running Asterisk version 13.7.1. I’ve configured several extensions and a sip trunk. All extensions can call each other further more its possible to make outgoing calls.
Unfortunatelly I’m not able to receive incoming calls. The extensions are configured as chan_sip on port 5060 as well as the trunk.
I’ve read several threads regarding call issues but unfortunatelly I haven’t found a solution for my issues. Could someone explain how to debug such a issue correctly?

What I’ve already done. Double checked the extension and trunk configuration. Trunk configuration is from a blog related to my provider shich should work. At Generals SIP settings I’ve set NAT to yes and IP Configuration to static IP since I obtain a static one from my provider.

To determine the issue I gone to the Asterisk CLI and set debug on. Unfortunatelly I’m not able to finde the issues within the output.
Is it correct that the call from my cell phone is registrated as “Unknown” and the FreePBX server tries to forward it to the extension and for some reaosn the extension rejects the call?

`
Asterisk 13.7.1, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 13.7.1 currently running on VoiP-Server (pid = 1634)
Reliably Transmitting (no NAT) to 172.16.0.10:5060:
OPTIONS sip:[email protected];line=6852 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK36a9dd40
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as50a73c6b
To: sip:[email protected];line=6852
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:172.16.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK36a9dd40
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as50a73c6b
To: sip:[email protected];line=6852
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Tue, 07 Jun 2016 08:49:29 GMT
Supported: replaces, timer
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 172.16.0.10:5060:
OPTIONS sip:[email protected];line=9032 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK60fe3c02
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as223ebbfe
To: sip:[email protected];line=9032
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:172.16.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK60fe3c02
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as223ebbfe
To: sip:[email protected];line=9032
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Tue, 07 Jun 2016 08:49:29 GMT
Supported: replaces, timer
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 217.0.23.100:5060:
OPTIONS sip:tel.myprovider.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK27143365;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as25bbe6f5
To: sip:tel.myprovider.com
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:217.0.23.100:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;rport=1365;branch=z9hG4bK27143365
To: sip:tel.myprovider.com;tag=h7g4Esbg_cy8v8vz72eib14dk8o4v9bmcpg4iu9fz
From: “Unknown” sip:[email protected];tag=as25bbe6f5
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:172.16.0.10:5060 —>
REGISTER sip:172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;rport;branch=z9hG4bKj8aori78d2z7cnjkudcbrnyen9uhm
Max-Forwards: 70
From: sip:[email protected];tag=k5v0yr6qg84cjo
To: sip:[email protected]
Call-ID: c8thhtm2d1f8yl0r.mfozx2rs
CSeq: 22768 REGISTER
Contact: sip:[email protected];line=9032
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Allow-Events: talk,hold
Expires: 3600
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 172.16.0.10:5060 (NAT)
Sending to 172.16.0.10:5060 (NAT)

<— Transmitting (no NAT) to 172.16.0.10:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bKj8aori78d2z7cnjkudcbrnyen9uhm;received=172.16.0.10;rport=5060
From: sip:[email protected];tag=k5v0yr6qg84cjo
To: sip:[email protected];tag=as60738306
Call-ID: c8thhtm2d1f8yl0r.mfozx2rs
CSeq: 22768 REGISTER
Server: FPBX-13.0.124(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="39025d73"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘c8thhtm2d1f8yl0r.mfozx2rs’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:172.16.0.10:5060 —>
REGISTER sip:172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;rport;branch=z9hG4bKqfzlnv1mow5v
Max-Forwards: 70
From: sip:[email protected];tag=k5v0yr6qg84cjo
To: sip:[email protected]
Call-ID: c8thhtm2d1f8yl0r.mfozx2rs
CSeq: 22769 REGISTER
Contact: sip:[email protected];line=9032
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Allow-Events: talk,hold
Authorization: Digest username=“109”, realm=“asterisk”, nonce=“39025d73”, uri=“sip:172.16.0.2”, response=“a9221b1e287ed56be193c8ed9f246d30”, algorithm=MD5
Expires: 3600
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 172.16.0.10:5060 (no NAT)
Reliably Transmitting (no NAT) to 172.16.0.10:5060:
OPTIONS sip:[email protected];line=9032 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK31edbfb5
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4e426b1f
To: sip:[email protected];line=9032
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— Transmitting (no NAT) to 172.16.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bKqfzlnv1mow5v;received=172.16.0.10;rport=5060
From: sip:[email protected];tag=k5v0yr6qg84cjo
To: sip:[email protected];tag=as60738306
Call-ID: c8thhtm2d1f8yl0r.mfozx2rs
CSeq: 22769 REGISTER
Server: FPBX-13.0.124(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: sip:[email protected];line=9032;expires=3600
Date: Tue, 07 Jun 2016 08:49:43 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘c8thhtm2d1f8yl0r.mfozx2rs’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:172.16.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK31edbfb5
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4e426b1f
To: sip:[email protected];line=9032
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Tue, 07 Jun 2016 08:49:43 GMT
Supported: replaces, timer
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:172.16.0.10:5060 —>
REGISTER sip:172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;rport;branch=z9hG4bKespxeft5vb7duiv86ifpftrst.yzs3
Max-Forwards: 70
From: sip:[email protected];tag=a0r2ag
To: sip:[email protected]
Call-ID: h67dc3yt158hmmxrir8e3.gg.is9y0
CSeq: 9063 REGISTER
Contact: sip:[email protected];line=6852
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Allow-Events: talk,hold
Expires: 3600
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 172.16.0.10:5060 (NAT)
Sending to 172.16.0.10:5060 (NAT)

<— Transmitting (no NAT) to 172.16.0.10:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bKespxeft5vb7duiv86ifpftrst.yzs3;received=172.16.0.10;rport=5060
From: sip:[email protected];tag=a0r2ag
To: sip:[email protected];tag=as65858164
Call-ID: h67dc3yt158hmmxrir8e3.gg.is9y0
CSeq: 9063 REGISTER
Server: FPBX-13.0.124(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7e728470"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘h67dc3yt158hmmxrir8e3.gg.is9y0’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:172.16.0.10:5060 —>
REGISTER sip:172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;rport;branch=z9hG4bKr2z5.tadivu.gn
Max-Forwards: 70
From: sip:[email protected];tag=a0r2ag
To: sip:[email protected]
Call-ID: h67dc3yt158hmmxrir8e3.gg.is9y0
CSeq: 9064 REGISTER
Contact: sip:[email protected];line=6852
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Allow-Events: talk,hold
Authorization: Digest username=“108”, realm=“asterisk”, nonce=“7e728470”, uri=“sip:172.16.0.2”, response=“74b05b30001795f408ee6ba1899a59f9”, algorithm=MD5
Expires: 3600
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 172.16.0.10:5060 (no NAT)
Reliably Transmitting (no NAT) to 172.16.0.10:5060:
OPTIONS sip:[email protected];line=6852 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK19091641
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4891b887
To: sip:[email protected];line=6852
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.124(13.7.1)
Date: Tue, 07 Jun 2016 08:49:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— Transmitting (no NAT) to 172.16.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bKr2z5.tadivu.gn;received=172.16.0.10;rport=5060
From: sip:[email protected];tag=a0r2ag
To: sip:[email protected];tag=as65858164
Call-ID: h67dc3yt158hmmxrir8e3.gg.is9y0
CSeq: 9064 REGISTER
Server: FPBX-13.0.124(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: sip:[email protected];line=6852;expires=3600
Date: Tue, 07 Jun 2016 08:49:45 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘h67dc3yt158hmmxrir8e3.gg.is9y0’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:172.16.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.2:5060;branch=z9hG4bK19091641
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4891b887
To: sip:[email protected];line=6852
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Tue, 07 Jun 2016 08:49:45 GMT
Supported: replaces, timer
User-Agent: Phone/03.24.0012 (MAC=000000000000; SER= 00000; HW=255)
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS`

Can some one point in the right direct or provide a way to debug this correctly?

Best regards,
Sablapet

I Couldn’t see any INVITE message. these are only REGISTER requests.

First make sure your SIP trunk is registered properly by using below commands from Asterisk CLI:
sip show peers
sip show subscription

Check if your inbound rule is configured properly and you are accepting calls from all CallerID, you can keep this field blank if you are not aware what caller ID you will receive for inbound calls.

Dear piyushaghera, psdk,

thanks for your answers.
@piyushaghera
My sip seems to be registered as well as the extensions. I assume I won’t be able to make outgoing calls if theres an issue. Nevertheless I’ll post the output of the show commands. (sip show subscription isn’t available on my system assume you mean sip show registry).:

<CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time tel.myprovider.com:5060 Y ***my-number*** 465 Registered Tue, 07 Jun 2016 21:47:19 1 SIP registrations.

<CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 108/108 172.16.0.10 D No No A 5060 OK (19 ms) 109/109 172.16.0.10 D No No A 5060 OK (19 ms) my-sip-trunk xxx.xxx.xxx.xxx Yes Yes 5060 OK (39 ms)

The extensions are registered at a IP-Dect base that why they have the same IP.

A inbound route is configured (as well as a outbound route):


Nearly default values. The other tabs arn’t changed.

@psdk
Thanks for indicating that a Invite is missing. Does this mean that the issue isn’t caused by the FreePBX server itself? Which mean the issue is within my network or my internet gateway? Is it possible to debug missing incoming packages with tcpdump or any equivalent tool?

I would like to add to my initial description. I’ve tried to do the test call with my cell phone. When I call my number I got no sound feedback like a ringtone or so on further more it takes about one minute and the call ended by itself. This seems to me that this is caused by a connection timeout by my cell phone provider.

Best regards,
Sablapet

I think you should talk with your provider!!! Because the trunk is registered but not any call request doesn’t come from it.

Simpler than that - set up an “any/any” incoming route with nothing in the DID or CID.

After that, you should see messages in your logs that say “incoming call with DID = something” in your logs. That’s your DID, not what you typed in the DID box on your incoming route.

Dear @cynjut thanks for your suggestion, I’ve done so but I can’t recognize any incoming calls. I’ve read the RFC 3665 to better understand how a incoming call should look like when it’s successful. And there is no INVITE message.

Dear @psdk I’ve called my provider and they confirmed that they aren’t able to call my number and furthermore confirmed the the trunk is registered correctly. Due to their standardized support process and since I’m not using one of the routers which are offered by them they cant provide further support for my issue. This is realy anoying but I’ve to accept this at this moment.

After this I’ve tried to use pjsip instead of chan_sip so I’ reconfigured my FreePBX installation, deleted the trunk, inbound as well as outbound route. I’ve disabled chan_sip and actived pjsip using port 5060. I’ve reconfigured my extensions to use pjsip, created a pjsip trunk (using a tutorial which is related to my provider) and created a new inbound and outbound route which is related to the new trunk.
After this I’m still able to make outgoing calls but not able to receive incoming calls.
I’ve logged the pjsip messages using the CLI and I already got no INVITE message.
What I’ve recognized is that I’ve got constantly a 403 Forbidden every 60 seconds:

`<— Transmitting SIP request (436 bytes) to UDP:MY_PROVIDER_IP:5060 —>
OPTIONS sip:MY_TELEPHONE_NUMBER@tel.my-provider.com:5060 SIP/2.0
Via: SIP/2.0/UDP MY_PUBLIC_IP:5060;rport;branch=z9hG4bKPjV9t4z5GXqVndRF91Dzfw0lPtsYu9DqcK
From: sip:[email protected]***FREEPBX_SERVER_IP***;tag=Ua1Aif4Tk7-eYj2N6bWPHau3vpTHIIQQ
To: sip:***MY_TELEPHONE_NUMBER***@tel.my-provider.com
Contact: sip:[email protected]:5060***MY_PUBLIC_IP***
Call-ID: pIc.WKe-wCBe7vWP6ahGx2w6moXD-ExJ
CSeq: 30669 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-13.0.119(13.7.1)
Content-Length: 0

<— Received SIP response (386 bytes) from UDP:MY_PROVIDER_IP:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY_PUBLIC_IP:5060;received=MY_PUBLIC_IP;rport=1027;branch=z9hG4bKPjV9t4z5GXqVndRF91Dzfw0lPtsYu9DqcK
To: sip:***MY_TELEPHONE_NUMBER***@tel.my-provider.com;tag=h7g4Esbg_zr2ouk3c9dxj32jjl2q2vshzm884q3hc
From: sip:[email protected]:5060***FREEPBX_SERVER_IP***;tag=Ua1Aif4Tk7-eYj2N6bWPHau3vpTHIIQQ
Call-ID: pIc.WKe-wCBe7vWP6ahGx2w6moXD-ExJ
CSeq: 30669 OPTIONS
Content-Length: 0`

Unfortunatelly I don’t understand what this sip message represents but I understand that the 403 respond from my provider means that the request is rejected.

Does anyone know what this message mean?

Best regards,
Sablapet

Is there any VoIP filter on your router? can you monitor traffics on your router interface?

Ps: It’s not far away that your provider want to sell a device to you!!!

Connect to the asterisk cli, give the following commands and make an inbound call, post here the output

core set verbose 5
core set debug 1

Your provider has told you that you are forbidden from using their server. From Wikipedia:

403 Forbidden - The server understood the request, but is refusing to fulfill it.

So, there is something wrong with your request. Since you are connecting to your own server, the error could easily be coming from your own machine.

Check the file /var/log/asterisk/full to see what the PBX is actually doing to process the call. SIP debug is just one of the tools you need.

Dear all,

thank you very much for your support.
I tooks several hours to find the issue but it seems to be fixed.
The Free PBX is part of its own “voip” vlan, which was managed by a switch. After recognizing that no packets arrive at the Free PBX server I took the time and read severals manuals, how to’s and command references regarding my network devices. I’ve investigated that the sip packets arrive at the router but are routed to a none existing ip. Afterwards I’ve decided to reconfigure my network topology and manage the vlans by the router itself. Since then I can recive incoming calls.
Additional I had a one way audio issue but I got it fixed by configuring the firewall (gateway).

Thank you very much.

Best regards,
Sablapet