- Joined
- Oct 22, 2015
- Messages
- 36
- Reaction score
- 1
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
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