Solved Incoming calls ring but get dropped on answer

[neilhooper]

Hello All,

I'm totally confused.

All my incoming calls get dropped immediately when answered even if left to go to voicemail.

I have looked in the logs but to honest, I don't know what I'm looking for however I did just notice this entry that I haven't seen before or maybe I've missed them previously.

20/04/2022 23:14:42 - [CM102001]: Authentication failed for AuthFail Recv Req REGISTER from 178.175.134.189:52391 tid=1044477372 Call-ID=282698384-1013321809-894053176: REGISTER sip:<my-ext-ip> SIP/2.0
Via: SIP/2.0/UDP 10.124.118.53:52391;branch=z9hG4bK1044477372;received=178.175.134.189
Max-Forwards: 70
Contact: <sip:[email protected]:52391>
To: <sip:200@<my-ext-ip>
From: <sip:200@<my-ext-ip> >;tag=1713260610
Call-ID: 282698384-1013321809-894053176
CSeq: 2 REGISTER
Proxy-Authorization: Digest
username="200,uri="sip:<my-ext-ip>",",algorithm=MD5,realm="3CXPhoneSystem",nonce="414d5359626085d122:c31e05be34c6aa27184cf518354e4105",response="e8ae31c949ce132ae0f964ac5e723801"
User-Agent: PolycomVVX-VVX_300-UA/4.1.6.4835
Content-Length: 0

; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

I don't know if this has anything to do with it.

I'm running Ver 18.0 Update 2 (Build 314) on a Rapsberry Pi
Our sip provider is Voicehost (UK) 3cx Supported.

Any help would be appreciated.

Neil

 

[cobaltit]

Does your system pass the firewall test?

 

[Saqqara]

Did you follow the setup https://www.3cx.com/docs/voicehost-uk-sip-trunk/

 

[leejor]

I suspect the log you posted is a hack attempt. One big clue is that it is directed to your public IP as opposed to your FQDN. You'd need to find a log of an incoming call from your provider.

 

[ChrisC_3CX]

If the call is ended after it has been answered then you'll probably see a SIP BYE message from either the PBX or SIP Provider and hence determine who is ending the call.

• Try making an incoming call to the PBX.
• Go to Dashboard >> Activity Log
• Click on the Filter button and select the last call (Provided the call you just placed is still the last one)
• Go through it and look for any BYE messages (you can use CTRL+F)

 

[neilhooper]

Does your system pass the firewall test?

Yes it does

Did you follow the setup https://www.3cx.com/docs/voicehost-uk-sip-trunk/

Yes I did

I've done another test call from my mobile phone to my number which when answered produces the following:

21/04/2022 20:23:40:688 | [Flow] Looking for inbound target: called=<my-incoming-number>; caller="Neil" <sip:<my-mobile-number>@:0>
21/04/2022 20:23:40:688 | [CM503012]: Inbound out-of-office hours rule (VH) for 10003 forwards to DN:16
21/04/2022 20:23:40:696 | Line limit check: Current # of calls for line Lc:10003(@VoiceHost[<sip:[email protected]:0/UDP>]) is 1; limit is 10
21/04/2022 20:23:40:697 | [CM503001]: Call(C:15): Incoming call from Line:10003<<<my-mobile-number> to <sip:[email protected]:5060>
21/04/2022 20:23:40:697 | [CM500002]: Call(C:15): Info on incoming INVITE from Line:10003<<<my-mobile-number>:
Invite-IN Recv Req INVITE from 185.91.41.29:5060 tid=Pjbe48e0fe-23ee-45ad-a21e-a8708c230e7c Call-ID=eb481a0d-3689-4bfc-b4ce-5b2cda15c1c9:
INVITE sip:<my-incoming-number>@<my-ext-ip>:5060;rinstance=b4e9013e111624b8 SIP/2.0
Via: SIP/2.0/UDP 185.91.41.29:5060;rport=5060;branch=z9hG4bKPjbe48e0fe-23ee-45ad-a21e-a8708c230e7c
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:<my-incoming-number>@<my-ext-ip>;rinstance=b4e9013e111624b8>
From: "<my-mobile-number>" <sip:<my-mobile-number>@185.91.41.29>;tag=308e783b-d06b-4bc3-88ed-a98eb8180983
Call-ID: eb481a0d-3689-4bfc-b4ce-5b2cda15c1c9
CSeq: 15236 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Supported: 100rel, timer, replaces, norefersub, histinfo
User-Agent: VoiceHost V6
Content-Length: 261

v=0
o=- 1725561653 1725561653 IN IP4 185.91.41.29
s=VoiceHost
c=IN IP4 185.91.41.29
t=0 0
m=audio 20066 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

21/04/2022 20:23:40:698 | [CM505003]: Provider:[VoiceHost] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:ST35205T001@<my-ext-ip>:5060]
21/04/2022 20:23:40:699 | [CM503010]: Call(C:15): Making route(s) from Line:10003<<<my-mobile-number> to <sip:[email protected]:5060/UDP>
21/04/2022 20:23:40:700 | [Flow] Target endpoint for 16 is Extn:16
21/04/2022 20:23:40:700 | [Flow] Call(C:15): has built target endpoint: Extn:16 for call from L:15.1[Line:10003<<<my-mobile-number>]
21/04/2022 20:23:40:700 | [CM503004]: Call(C:15): Route 1: from L:15.1[Line:10003<<<my-mobile-number>] to T:Extn:16@[Dev:sip:[email protected]:5062]
21/04/2022 20:23:40:700 | [CM503027]: Call(C:15): From: Line:10003<<<my-mobile-number> ("VH:Neil" <sip:<my-mobile-number>@<my-fdqn>:5060>) to T:Extn:16@[Dev:sip:[email protected]:5062]
21/04/2022 20:23:40:714 | [CM503025]: Call(C:15): Calling T:Extn:16@[Dev:sip:[email protected]:5062] for L:15.1[Line:10003<<<my-mobile-number>]
21/04/2022 20:23:40:881 | [CM503002]: Call(C:15): Alerting Extn:16 by contact <sip:[email protected]:5062/UDP>
21/04/2022 20:23:40:881 | [CM505001]: Endpoint Extn:16: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Yealink SIP-T26P 6.73.0.60] PBX contact: [sip:[email protected]:5060]
21/04/2022 20:23:45:441 | L:15.2[Extn:16] has joined to L:15.1[Line:10003<<<my-mobile-number>]
21/04/2022 20:23:45:453 | [CM503007]: Call(C:15): Extn:16 has joined, contact <sip:[email protected]:5062/UDP>
21/04/2022 20:23:45:456 | [CM503007]: Call(C:15): Line:10003<<<my-mobile-number> has joined, contact <sip:[email protected]:0/UDP>
21/04/2022 20:23:45:558 | Leg L:15.1[Line:10003<<<my-mobile-number>] is terminated: Cause: BYE from 185.91.41.29:5060
21/04/2022 20:23:45:559 | [CM503008]: Call(C:15): Call is terminated
21/04/2022 20:23:45:562 | Leg L:15.2[Extn:16] is terminated: Cause: BYE from local


I think the problem is with Voicehost as at 20:23:45:558 there is a BYE from 185.91.41.29:5060

The IP address range 185.91.41.xxx seems to belong to Voicehost so if my thinking is correct then I need to be speaking with them.

Please correct me if I'm wrong and thanks for the speedy replies.

 

[ChrisC_3CX]

I think the problem is with Voicehost as at 20:23:45:558 there is a BYE from 185.91.41.29:5060

The IP address range 185.91.41.xxx seems to belong to Voicehost so if my thinking is correct then I need to be speaking with them.

Please correct me if I'm wrong and thanks for the speedy replies.

That is absolutely correct, I would indeed recommend reaching out to them about this and see what they have to say.

 

[neilhooper]

Update.

It turned out to be the SRTP setting in SIPTrunks > "Sip Tunk Name" > Options, was set to enforced which meant that their end should have had RTS enabled but it wasn't. I changed my end to disabled and everything is fine.

Just thought I'd let you all know, it might help someone else one day.

 

[ChrisC_3CX]

Thank you for the update! I'm sure others with similar issues will find this helpful. I do have to mention though that our SIP Trunk template for Voicehost has SRTP disabled by default. That said, always make sure you're using the default 3CX template when utilizing a 3CX Supported Provider as it is with those settings the specific provider is tested and supported with 3CX.