Call is disconnected

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comfident

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Hi,

One call, from one specific caller, is not working (he even does not hear a ring tone).
The call simple end, when he calls the main number and if he calls an extensions, it rings about 3 times and the call is then also over.
Other calls from other callers just simply work normally.

This is was i get in the logs for calling the main number:

....
....
08/03/2023 11:26:21 AM - L:290.1[Line:10000<<+436601122333] got Connected.UAS Send 200/INVITE from 0.0.0.0:0 tid=-*10*-1-a9f229734bd564c2a911taN1 Call-ID=O3BElv0kwhLUBROA1Z0369cepU8_o-D1deba@zteims:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 194.24.130.128:5062;branch=z9hG4bK-*10*-1-a9f229734bd564c2a911taN1
Require: timer
Contact: <sip:[email protected]:5061;transport=TLS>
To: <sip:[email protected]>;tag=3cf7f306
From: <sip:[email protected]>;tag=ztesip0TE6eYgNBPSlMH558E-eoX*13-1-20481*ffhc.13
Call-ID: O3BElv0kwhLUBROA1Z0369cepU8_o-D1deba@zteims
CSeq: 1000 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
User-Agent: 3CXPhoneSystem 18.0.7.312 (312)
Content-Length: 306

v=0
o=3cxPS 3732501029715968 2890579125993473 IN IP4 222.111.222.111
s=3cxPS Audio call
c=IN IP4 222.111.222.111
t=0 0
m=audio 9208 RTP/SAVP 8 100
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:zwXuBIgqAJjEI4JggSodCm90zsRwMKOLll44afnj
a=sendrecv


08/03/2023 11:26:21 AM - [CM503007]: Call(C:290): Queue:803 has joined, contact <sip:[email protected]:5483/UDP>

08/03/2023 11:26:21 AM - [CM503007]: Call(C:290): Line:10000<<+436601122333 has joined, contact <sip:[email protected]:5062/UDP>

08/03/2023 11:26:21 AM - CA request makeCall(dn_from=, number_to=, handler=803)

08/03/2023 11:26:21 AM - MakeCall request is sent to 803

08/03/2023 11:26:22 AM - SendMsg from <sip:[email protected]>;tag=3cf7f306 to <sip:[email protected]>;tag=ztesip0TE6eYgNBPSlMH558E-eoX*13-1-20481*ffhc.13

08/03/2023 11:26:22 AM - L:290.1[Line:10000<<+436601122333] Sending: OnSendResp Send 200/BYE from 0.0.0.0:0 tid=-*10*-1-1302601cb958c952b53etaN1 Call-ID=O3BElv0kwhLUBROA1Z0369cepU8_o-D1deba@zteims:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 194.24.130.128:5062;branch=z9hG4bK-*10*-1-1302601cb958c952b53etaN1
Contact: <sip:[email protected]:5061;transport=TLS>
To: <sip:[email protected]>;tag=3cf7f306
From: <sip:[email protected]>;tag=ztesip0TE6eYgNBPSlMH558E-eoX*13-1-20481*ffhc.13
Call-ID: O3BElv0kwhLUBROA1Z0369cepU8_o-D1deba@zteims
CSeq: 1001 BYE
Content-Length: 0

08/03/2023 11:26:22 AM - Terminated from <sip:[email protected]>;tag=ztesip0TE6eYgNBPSlMH558E-eoX*13-1-20481*ffhc.13 to <sip:[email protected]>;tag=3cf7f306; reason: RemoteBye
...

Can someone see a reason in the logs, which side is terminating the call and why?

Regards
 
since all other calls work and the issue is only with one number the best is to contact your provider.
 
Hi,
Can you see in the posted logs, which side is terminating the call and why?
 
some logs seem to be missing, from what I can see, the call is terminating from the providers side.
 
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