Rolling calls to answering service

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CrashDodson

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We upgraded to the latest version and also moved from a sip trunk provided by AT&T (ipflex) to a "local" provider that's in our same datacenter. They are not on the approved list so 3cx support won't assist with the issue. We have a ring group that rolls for a few seconds then if no answer we send calls to an outside number/answering service. Been doing this for years prior to the upgrade and sip trunk change. After the timeout it's like the call just rings back into the system. It will do this until all the licenses are consumed. We have tried sending the call directly out of the ring group to the answering service and forwarding the call to a dummy extension that is forwarded to the answering service with the same result.
 
After the timeout it's like the call just rings back into the system. It will do this until all the licenses are consumed.
Use the 3CX Activity Log (Verbose) to see exactly where the call is routing. It sounds as if you have loop somewhere. Confirm that if the call is indeed going to the answering service, that they are not sending it back to you.
 
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So I have a did 432242**** I am sending to an unregistered extension 1097. If the forwarding is set to voicemail it works fine. If I try to forward it to a mobile of 432312**** the call fails. Im calling from 432770**** in the logs.

We tested this on another system. Same version of 3cx, the same SIP trunk provider and it works. I have tried changing various options on the trunk with no change. When i call from my cell I get a message that says "Your call cannot be completed at this time, please try again later". Which sounds like a telco recording.

05/08/2023 4:44:06 PM - ~Call(C:325)
05/08/2023 4:44:06 PM - ~CallLeg(C:325.1)
05/08/2023 4:44:06 PM - Call(325): Device (1185951132) is used in 0 calls: []
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****]: destroying InvADS.ADS(17272)
05/08/2023 4:44:06 PM - Call(C:325) has been terminated already
05/08/2023 4:44:06 PM - C:325: termination check: the call has 0 legs
05/08/2023 4:44:06 PM - Call(C:325): Legs count: 0
05/08/2023 4:44:06 PM - Stop call record for leg L:325.1[Line:10000<<432770****]
05/08/2023 4:44:06 PM - Removing LegInfo[C:325.1/Terminating@GotOffer->0]
05/08/2023 4:44:06 PM - Removing leg L:325.1[Line:10000<<432770****]
05/08/2023 4:44:06 PM - Leg L:325.1[Line:10000<<432770****] is terminated: Cause: BYE from PBX
05/08/2023 4:44:06 PM - Ignored: SE:325.1{OnHook}
05/08/2023 4:44:06 PM - StatusNotification: SE:325.1{OnHook}; Skip=0
05/08/2023 4:44:06 PM - Call(325): Device (1185951132) is used in 0 calls: []
05/08/2023 4:44:06 PM - Changed LegInfo[C:325.1/Terminating@GotOffer->0]
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] Sending: OnSendResp Send 480/INVITE from 0.0.0.0:0 tid=1sansay1035266892rdb11670 Call-ID=[email protected]:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK1sansay1035266892rdb11670
To: <sip:432242****@1.1.1.1>;tag=b94b3b60
From: <sip:432770****@2.2.2.2>;tag=sansay1035266892rdb11670
Call-ID: [email protected]
CSeq: 1 INVITE
Warning: 499 Metering-3CX "Not available"
Content-Length: 0

05/08/2023 4:44:06 PM - Call(C:325) is terminated
05/08/2023 4:44:06 PM - Reported(Delete): SE:325.1{OnHook}
05/08/2023 4:44:06 PM - StatusNotification: SE:325.1{OnHook}; Skip=0
05/08/2023 4:44:06 PM - [CM503020]: Call(C:325): Normal call termination. Call originator: Line:10000<<432770****. Reason: Not available
05/08/2023 4:44:06 PM - [CM503016]: Call(C:325): Attempt to reach <sip:[email protected]:5060> from Line:10000<<432770**** has failed. Reason: Not Registered
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] failed to reach Out#:>>Rule{10 Digit}>>432312****, reason Not Registered
05/08/2023 4:44:06 PM - [CM503016]: Call(C:325): Attempt to reach <sip:[email protected]:5060> from Line:10000<<432770**** has failed. Reason: Not Registered
05/08/2023 4:44:06 PM - [CM503017]: Call(C:325): Target is not registered: Out#:>>Rule{10 Digit}>>432312****
05/08/2023 4:44:06 PM - Leg L:325.1[Line:10000<<432770****]: Refreshing OutCID from endpoint Extn:1097 = [432242****]
05/08/2023 4:44:06 PM - [Flow] Call(C:325): applied forwarding rule (Fwd[Away/AllCalls]) Extn:1097 -> Out#:>>Rule{10 Digit}>>432312****
05/08/2023 4:44:06 PM - [Flow] Call(C:325): has built target endpoint: Extn:1097 for call from L:325.1[Line:10000<<432770****]
05/08/2023 4:44:06 PM - [CM503010]: Call(C:325): Making route(s) from Line:10000<<432770**** to <sip:[email protected]:5060/UDP>
05/08/2023 4:44:06 PM - Remote SDP is set for leg L:325.1[Line:10000<<432770****]
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] set remote SDP (Offer):
v=0
o=Sansay-VSXi 188 1 IN IP4 2.2.2.2
s=Session Controller
c=IN IP4 74.120.82.34
t=0 0
m=audio 32470 RTP/AVP 0 18 100
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=ptime:20
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] request to join MSCall [email protected]:5482
05/08/2023 4:44:06 PM - LegInfo[C:325.1/CallIn@None->0] SDP state change request => GotOffer {612}
05/08/2023 4:44:06 PM - LegInfo[C:325.1/CallIn@None->0] setting remote offer. SDP:
v=0
o=Sansay-VSXi 188 1 IN IP4 2.2.2.2
s=Session Controller
c=IN IP4 74.120.82.34
t=0 0
m=audio 32470 RTP/AVP 0 18 100
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=ptime:20
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] got remote SDP offer
05/08/2023 4:44:06 PM - OnOffer from L:325.1[Line:10000<<432770****]
05/08/2023 4:44:06 PM - Reported(Insert): SE:325.1{Calling}
05/08/2023 4:44:06 PM - StatusNotification: SE:325.1{Calling}; Skip=0
05/08/2023 4:44:06 PM - [CM500002]: Call(C:325): Info on incoming INVITE from Line:10000<<432770****:
Invite-IN Recv Req INVITE from 2.2.2.2:5060 tid=1sansay1035266892rdb11670 Call-ID=[email protected]:
INVITE sip:432242****@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK1sansay1035266892rdb11670
Max-Forwards: 69
Record-Route: <sip:[email protected]:5060;lr;transport=udp>
Contact: <sip:432770****@2.2.2.2:5060>
To: <sip:432242****@1.1.1.1>
From: <sip:432770****@2.2.2.2>;tag=sansay1035266892rdb11670
Call-ID: [email protected]
CSeq: 1 INVITE
Session-Expires: 600;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Supported: timer, 100rel, replaces
P-Asserted-Identity: <sip:432770****;verstat=[email protected]:5060>
Content-Length: 272

v=0
o=Sansay-VSXi 188 1 IN IP4 2.2.2.2
s=Session Controller
c=IN IP4 74.120.82.34
t=0 0
m=audio 32470 RTP/AVP 0 18 100
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=ptime:20
05/08/2023 4:44:06 PM - [CM503001]: Call(C:325): Incoming call from Line:10000<<432770**** to <sip:[email protected]:5060>
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] NoSDP - false
05/08/2023 4:44:06 PM - Leg L:325.1[Line:10000<<432770****] has external party ID {432770****}
05/08/2023 4:44:06 PM - NAT/ALG check:L:325.1[Line:10000<<432770****] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop
Media session IP ('c=' attribute) is not equal to the IP specified in contact header:
Media session IP:74.120.82.34
Contact IP:2.2.2.2
Media session IP ('c=' attribute) is not equal to the SIP packet source(IP:port):
Media session IP: 74.120.82.34
Received from: 2.2.2.2

05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] joined to MSCall([email protected]:5482)
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] request to join MSCall [email protected]:5482
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] created MSEP ([email protected]:5482) (bound)
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****]: forced source is used for outbound: 2.2.2.2
05/08/2023 4:44:06 PM - Call(325): Device (1185951132) is used in 1 calls: [325]
05/08/2023 4:44:06 PM - IncomingCall: C:325 from <sip:432770****@2.2.2.2:0/UDP> to <sip:432242****@1.1.1.1:5060/UDP>
05/08/2023 4:44:06 PM - Added leg L:C:325.1[No endpoint yet]
05/08/2023 4:44:06 PM - Added LegInfo[C:325.1/CallIn@None->0]
 
Think I may have figured it out. Doing some more testing.
 
Did you figure it out?

If not, please try to add the outside number in your Outbound rule (as a number), place this rule at the top, and let the number transferring the call be able to use this rule, then I think you are good to go
 
In the log you posted, it shows the call being terminated by the PBX because the number was unavailable.
05/08/2023 4:44:06 PM - Call(C:325) is terminated
05/08/2023 4:44:06 PM - Reported(Delete): SE:325.1{OnHook}
05/08/2023 4:44:06 PM - StatusNotification: SE:325.1{OnHook}; Skip=0
05/08/2023 4:44:06 PM - [CM503020]: Call(C:325): Normal call termination. Call originator: Line:10000<<432770****. Reason: Not available
05/08/2023 4:44:06 PM - [CM503016]: Call(C:325): Attempt to reach <sip:[email protected]:5060> from Line:10000<<432770**** has failed. Reason: Not Registered
05/08/2023 4:44:06 PM - L:325.1[Line:10000<<432770****] failed to reach Out#:>>Rule{10 Digit}>>432312****, reason Not Registered
05/08/2023 4:44:06 PM - [CM503016]: Call(C:325): Attempt to reach <sip:[email protected]:5060> from Line:10000<<432770**** has failed. Reason: Not Registered
Are you using the extension forwarding options for it when not registered?

https://www.3cx.com/community/threa...gistered-not-ring-mobile-simultaneous.117038/
 
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