Strange Issue: Call Queue / Destination if no answer not working for external calls

[LH74]

Hi there,

I've set up a Queue with 2 Agents. If both Agents are logged out the "Destination if no answer" should get the call. This works on internal calls only. External Calls are terminated after 2 seconds. Any idea for troubleshooting?

This is, what I've tried already:
* Assign a DID for this Queue for test, same behaviour when directly called this DID
* Changed No-Answer-Destination from Voicemail to Extension or external number, same behaviour

 

[LH74]

Here is the Activity Log:

02/12/2020 9:51:48 PM - Leg L:106.1[Line:10001<<017610000000] is terminated: Cause: BYE from local
02/12/2020 9:51:48 PM - [CM503008]: Call(C:106): Call is terminated
02/12/2020 9:51:48 PM - Leg L:106.2[Queue:803] is terminated: Cause: BYE from local
02/12/2020 9:51:46 PM - [CM503007]: Call(C:106): Queue:803 has joined, contact <sip:[email protected]:5483/UDP>
02/12/2020 9:51:46 PM - [CM503007]: Call(C:106): Line:10001<<017610000000 has joined, contact <sip:[email protected]:0/UDP>
02/12/2020 9:51:46 PM - L:106.2[Queue:803] has joined to L:106.1[Line:10001<<017610000000]
02/12/2020 9:51:46 PM - [CM505001]: Endpoint Queue:803: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Queue] PBX contact: [sip:[email protected]:5060]
02/12/2020 9:51:46 PM - [CM503002]: Call(C:106): Alerting Queue:803 by contact <sip:[email protected]:5483/UDP>
02/12/2020 9:51:46 PM - [CM503025]: Call(C:106): Calling T:Queue:803@[Dev:sip:[email protected]:5483;rinstance=fa7281fd10f7f33c] for L:106.1[Line:10001<<017610000000]
02/12/2020 9:51:46 PM - [CM503027]: Call(C:106): From: Line:10001<<017610000000 (<sip:[email protected]3cx.de:5060>) to T:Queue:803@[Dev:sip:[email protected]:5483;rinstance=fa7281fd10f7f33c]
02/12/2020 9:51:46 PM - [CM503004]: Call(C:106): Route 1: from L:106.1[Line:10001<<017610000000] to T:Queue:803@[Dev:sip:[email protected]:5483;rinstance=fa7281fd10f7f33c]
02/12/2020 9:51:46 PM - [Flow] Call(C:106): has built target endpoint: Queue:803 for call from L:106.1[Line:10001<<017610000000]
02/12/2020 9:51:46 PM - [Flow] Target endpoint for 803 is Queue:803
02/12/2020 9:51:46 PM - [CM503010]: Call(C:106): Making route(s) from Line:10001<<017610000000 to <sip:[email protected]:5060/UDP>
02/12/2020 9:51:46 PM - [CM505003]: Provider:[Easybell] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
02/12/2020 9:51:46 PM - [CM500002]: Call(C:106): Info on incoming INVITE from Line:10001<<017610000000:
Invite-IN Recv Req INVITE from 195.185.37.60:5060 tid=Y9Qo~amqiwkQs Call-ID=SBCe615e00009f5-5e446561-6706d8c6-3a5b3d50-5f198ea-01_b2b-1:
INVITE sip:[email protected]:5060;rinstance=5977d7fbf012480b SIP/2.0
Via: SIP/2.0/UDP 195.185.37.60;branch=z9hG4bKY9Qo~amqiwkQs;rport=5060
Max-Forwards: 68
Contact: <sip:[email protected];transport=udp>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=12F04C07-5E44656200014862-998E3700
Call-ID: SBCe615e00009f5-5e446561-6706d8c6-3a5b3d50-5f198ea-01_b2b-1
CSeq: 10 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK
Content-Type: application/sdp
Supported: histinfo
Content-Length: 243

v=0
o=- 0 0 IN IP4 195.185.37.60
s=-
c=IN IP4 195.185.37.60
t=0 0
m=audio 25818 RTP/AVP 8 0 103
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:103 telephone-event/8000
a=fmtp:103 0-15
a=direction:both
a=sendrecv
a=maxptime:20
02/12/2020 9:51:46 PM - [CM503001]: Call(C:106): Incoming call from Line:10001<<017610000000 to <sip:[email protected]:5060>
02/12/2020 9:51:46 PM - Line limit check: Current # of calls for line Lc:10001(@easybell[<sip:[email protected]:0/UDP>]) is 1; limit is 10
02/12/2020 9:51:46 PM - Timezone offset: 60 min.

 

[LH74]

Here's another follow up on this issue:

If I call from my pbx (over PSTN), everything is fine
If I call from another landline, everythin is fine
If I call from mobile Network, some calls are terminated successful, but mostly not

It seems, the issue is not caused by the Queue Settings, but Maybe in the SIP-Trunk. Obviously Problems occour from mobile Network only.

 

[LH74]

Seems to be solved: I've disabled all the Codecs except G.711 U-law. Now calls from mobile-Network gets no longer disconnected.