Unsupported Incoming call rejected, caller is unknown

[NoFate]

Hey,

Testing an on premise solution
created 3 extensions
1000 => video intercom
1001 => linphone client 1
1002 => linphone client 2

from 1000 i can call to 1001 and visa versa
now, i want to dial out from my intrecom 1000 to 1001 , but gives error below
what is the cause?

its all internal on same subnet, no inboud rule is created, no sip trunk ....

thnx in advance

09/29/2021 3:47:52 PM - [CM503013]: Call(C:21): Incoming call rejected, caller is unknown; msg=Invite-IN Recv Req INVITE from 192.168.0.70:5060 tid=1486370746 [email protected]: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.70:5060;rport=5060;branch=z9hG4bK1486370746 Max-Forwards: 20 Contact: <sip:[email protected]:5060> To: <sip:[email protected]:5060> From: "1000"<sip:[email protected]>;tag=1546765310 Call-ID: [email protected] CSeq: 62 INVITE Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1000",realm="3CXPhoneSystem",nonce="414d535961546e8750:e658bd5e38e7d48b1f3992785fde20d4",uri="sip:[email protected]:5060",response="caa1830a4fb9a710fc59fd08b8bc5ae4",algorithm=MD5 User-Agent: YATE/5.5.0 Content-Length: 288 v=0 o=yate 1632930470 1632930470 IN IP4 192.168.0.70 s=SIP Call c=IN IP4 192.168.0.70 t=0 0 m=audio 9654 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 9856 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

09/29/2021 3:47:52 PM - Leg L:21.1[Unknown:] is terminated: Cause: BYE from PBX

09/29/2021 3:47:51 PM - [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification: Invite-UNK Recv Req INVITE from 192.168.0.70:5060 tid=1821047193 [email protected]: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.70:5060;rport=5060;branch=z9hG4bK1821047193 Max-Forwards: 20 Contact: <sip:[email protected]:5060> To: <sip:[email protected]:5060> From: "1000" <sip:[email protected]>;tag=1546765310 Call-ID: [email protected] CSeq: 61 INVITE Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO Content-Type: application/sdp User-Agent: YATE/5.5.0 Content-Length: 288 v=0 o=yate 1632930470 1632930470 IN IP4 192.168.0.70 s=SIP Call c=IN IP4 192.168.0.70 t=0 0 m=audio 9654 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 9856 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

 

[Alphabetic]

I would suggest using a supported setup such as using the 3CX Clients.

 

[NoFate]

i cant use the 3cx client, thats why i was asking

 

[NoFate]

did a quick test , and installed 3cx on android, but no difference
it must be a setting at the caller orginator itself

 

[leejor]

from 1000 i can call to 1001 and visa versa
now, i want to dial out from my intrecom 1000 to 1001 , but gives error below
what is the cause?

Do you mean you want to call from 1000 to an outside trunk?

I have not seem an issue previously, where, if an extension can call internally, then there is a source identification issue, when attempting an outside call. That sort of issue usually prevent all calls from being processed.

Perhaps you can provide more information about your setup...
https://www.3cx.com/community/threads/information-to-provide-when-requesting-help.67558/

 

[NoFate]

no, all calls are internal
all on same local subnet

its a verry basic setup, (on premise) afrer first install wizard, just created 3 extensions/users ... thats about it

now just registered those extensions...

no additional config at all

 

[leejor]

Now I'm a bit confused. In the first line you say

from 1000 i can call to 1001 and visa versa

Did you mean to say that you can't call from 1000 to 1001?

Did you do a manual setup on Ext 1000?
Does the extension appear in 3Cx as registered?
Does Ext 1000 actually receive calls?

 

[NoFate]

indeed,
i cant call from from 1000 (intercom) to any other SIP client
allthough 1000 shows up as regstered
i tested before asterisk/freepbx/yeastar, and that worked
calling to 1000 doesnt work, its an intercom, it doesnt have an answer button
but calling from 1000 to other sip clients should work

i always see this in log :

[CM500002]: Unidentified incoming call. Review INVITE and adjust source identification: Invite-UNK Recv Req INVITE from
and then
[CM503013]: Call(C:21): Incoming call rejected, caller is unknown

not sure why... something is happening when i call from 1000 to .... but 3cx is rejecting it?
i also whitelisted my local lan 192.168.0.0/24 , doesnt help either

there is no trunk setup, not needed, no rules/routing ....

 

[NoFate]

trying to change the first post , but seems there is no edit button there
indeed, i cant call from 1000 to1001 ot 1002

i can call from 1001 to 1002 or 1002 to 1001 (regular sip clients)
so that means 3cx is working fine

its just from 1000 (intrecom) to any other clients is being rejected

 

[NoFate]

here is the log when it registers fine :

09/29/2021 9:52:25 PM - [CM504001]: Endpoint Extn:1000: new contact is registered. Contact(s): [sip:[email protected]:5060 / 1000]

if i compare that to the log where i see the "caller unknown" i have no idea why, ip is identical

 

[NoFate]

 

[NoFate]

anyone that can help me? what could be the cause?

 

[NoFate]

No one that can help with this?