calls to ring group don't go to VM

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Leon Zetekoff

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currently have one DID coming into a RG and that rings three extensions. The call is coming in from my old PBX (temporarily till I get everything migrated). The extensions ring but if someone REJECTS the call it goes to the VM of a cell phone (which is ext 701s cell #). Instead of the 709 which is defined in the RG.

If he doesnt reject the call on timeout the call reverts back to the IVR on the old PBX.

if I dial in from a random phone to the New PBX and it times out it goes to that same VM 701s cell phone,

I am stumped :-(

Thanks leon

RG 800
709 701 715
timeout --> 709 VM

cell phones defines on 701 and 715 not on 709 since its in the UK
 
ok if I dial 709 from 715 and let it timeout it goes to 709 VM
if I dial 800 from 715 and let it timeout it goes to 709 VM
if I dial into the pbx from another number the ring group rings but times out to cellphone

06/03/2018 3:51:06 PM - Currently active calls [none]
06/03/2018 3:50:38 PM - Leg L:37.1[EndCall:EndCall] is terminated: Cause: BYE from PBX
06/03/2018 3:50:37 PM - Source identification failed: Caller is not identified
06/03/2018 3:50:37 PM - [CM503013]: Call(C:37): Incoming call rejected, caller is unknown; msg=Invite-IN Recv Req INVITE from 72.251.239.205:41726 tid=56612a19 [email protected]:5060: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/TCP 72.251.239.205:5060;branch=z9hG4bK56612a19 Max-Forwards: 70 Contact: <sip:[email protected]:5060;transport=TCP> To: <sip:[email protected]:5060> From: "BackWoods Wireless" <sip:[email protected]>;tag=as2d37a6cd Call-ID: [email protected]:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Content-Type: application/sdp Date: Sun, 03 Jun 2018 19:50:37 GMT Supported: replaces, timer User-Agent: voip.ms Remote-Party-ID: "BackWoods Wireless" <sip:[email protected]>;party=calling;privacy=off;screen=no Content-Length: 248 v=0 o=root 697454212 697454212 IN IP4 72.251.239.205 s=voip.ms c=IN IP4 72.251.239.205 t=0 0 m=audio 17956 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
06/03/2018 3:50:37 PM - Failed to identify source: several trunks matched by DID only. SI-Err Recv Req INVITE from 72.251.239.205:41726 tid=56612a19 [email protected]:5060: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/TCP 72.251.239.205:5060;branch=z9hG4bK56612a19 Max-Forwards: 70 Contact: <sip:[email protected]:5060;transport=TCP> To: <sip:[email protected]:5060> From: "BackWoods Wireless" <sip:[email protected]>;tag=as2d37a6cd Call-ID: [email protected]:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Content-Type: application/sdp Date: Sun, 03 Jun 2018 19:50:37 GMT Supported: replaces, timer User-Agent: voip.ms Remote-Party-ID: "BackWoods Wireless" <sip:[email protected]>;party=calling;privacy=off;screen=no Content-Length: 248 v=0 o=root 697454212 697454212 IN IP4 72.251.239.205 s=voip.ms c=IN IP4 72.251.239.205 t=0 0 m=audio 17956 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
06/03/2018 3:50:37 PM - More than one line have matched by DID only
06/03/2018 3:50:37 PM - The SIP request source identification procedure matches several trunks by DID onlySI-Err Recv Req INVITE from 72.251.239.205:41726 tid=56612a19 [email protected]:5060: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/TCP 72.251.239.205:5060;branch=z9hG4bK56612a19 Max-Forwards: 70 Contact: <sip:[email protected]:5060;transport=TCP> To: <sip:[email protected]:5060> From: "BackWoods Wireless" <sip:[email protected]>;tag=as2d37a6cd Call-ID: [email protected]:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Content-Type: application/sdp Date: Sun, 03 Jun 2018 19:50:37 GMT Supported: replaces, timer User-Agent: voip.ms Remote-Party-ID: "BackWoods Wireless" <sip:[email protected]>;party=calling;privacy=off;screen=no Content-Length: 248 v=0 o=root 697454212 697454212 IN IP4 72.251.239.205 s=voip.ms c=IN IP4 72.251.239.205 t=0 0 m=audio 17956 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
 
i took the RG out of the inbound route and works the same timeout goes to a cellphone VM
 
If the call destination (the one that answers), whether part of a ring group or not, is a mobile phone (forwarded from a 3CX extension), and the call is rejected by the mobile, then routing will follow the mobiles predetermined destination, which is probably it's voicemail.
I hope I am understanding the issue correctly.

A few concerns in the log..

Failed to identify source: several trunks matched by DID only.

Source identification failed: Caller is not identified
06/03/2018 3:50:37 PM - [CM503013]: Call(C:37): Incoming call rejected, caller is unknown;
 
Hi there...the cell phone has extension 701 on it using the 3cx client. so I still dont understand how it's getting there.

Also, if a device REJECTS the call the pbx should keep dialing the rest of the devices its dealing with. WHy is a REJECT from one device causing the ring to cancel?

Thanks leon

PS: Yeah I saw those entries in the log too and don't know where/why we're getting them?
 
and to be clear the call goes to the cells actual cell phone number NOT 701
 
Ok so the extension 701 is part of the Ring Group and has enabled the option "Ring my mobile simultaneously" enabled under the forwarding rules. Is that correct?

If the call is rejected by the actual cell phone number and the cell phone has destination set as voicemail then the call is answered by the cells phones voicemail and the PBX will stop polling the other agents as it will receive a 200 OK from the cell phone.
 
HI there....NO, it is NOT set to simultaneously ring cell phone. A REJECT from any device ringing should NOT cause it to go to that device VM or some other place. It should still follow the programmed actions.

the purpose of a ring group is to ring multiple devices and a reject from any one of them should not stop it unilaterallly.
 
The first step is to try and identify the issue. If you call the Ring Group for an extension of the 3CX PBX does it behave normally and polls all 3 extensions and goes to the correct destination if no answer?
Is the issue only with calls coming from the old system? How is the old system connected to the PBX? have you connected it as a trunk or through a bridge? Are you dialling an extension or a DID?
 
1. no. I will have to retest when I can let them know not to answer it. last I know the timeout went to a cell phone vm

2. no happens if I call in directly. If I call ext 709 from INSIDE it seems to work. I think if I called RG800 it failed too.
the old system is not connected at all. it is on a separate server on digital ocean. This is on vultr. I'm trying to migrate from old to new but stopped until this gets sorted out.

Everything is via POTS unless I said otherwise.
 
the purpose of a ring group is to ring multiple devices and a reject from any one of them should not stop it unilaterallly.

If I understand correctly, the issue boils down to the above. If a ring group member (a 3CX client on an Android device) "Rejects a ring group call, should the call go to... that extensions voicemail, or, to the "unanswered" RG destination, or, continue ringing other members of the ring group. I think that the first option is happening, but he is expecting the last. I think that rejecting a call, in a ring group (something a desk set really can't do), would appear to be treated as "answered", in a way, but then 3CX has to do something with the call, and it sounds as if that is to send it to the voicemail of the extension as it would with a normal, (not from a RG) call.
 
hi there....close....

call comes into DID goes to RG.
RG rings extensions. All extensions ring. TImeout should go to the destination set in the RG.
OR
All extensions ring. one of the extensions REJECT the call. Call should still ring the remaining destinations in RG and then follow any RG timeout set.

I will double check over the next day or so the current failure mode. GOt back too late tonight to do anything.

Thanks leon
 
RG rings extensions. All extensions ring. TImeout should go to the destination set in the RG.
OR
All extensions ring. one of the extensions REJECT the call. Call should still ring the remaining destinations in RG and then follow any RG timeout set.
That is the expected behaviour of a ring group. If the call times out it should forward to the destination if no answer of the Ring Group.
Also if a member of the ring group rejects the call the other members should continue to ring as long as the call is rejected by a 3CX extension.

Please check where this fails and let us know so we can focus on point of failure.
 
I'm puzzled by a couple of statements made earlier as they suggest other problems.



If he doesnt reject the call on timeout the call reverts back to the IVR on the old PBX.

This would seem to be a short time-out period set in the old PBXs IVR and the call is considered unanswered for too long.

and to be clear the call goes to the cells actual cell phone number NOT 701

So, somehow, the mobile users cell phone number IS involved‽ Forwarding, or Ring my Mobile must be happening to the ring group calls. If that is the case, and the call goes off to the mobile voicemail, then it is considered answered. If it goes to both the 3CX extension, on the mobile AND rings the mobiles PSTN number, then that is probably going to cause problems as you have two calls coming in on the same handset, at the same time.
 
I think I know whats happening. At voip.ms there are three failure settings which redirect the call to the Rabbi's cellphone. I predict this is what's happening. I never had this happen on Asterisk/freepbx unless the pbx was toast. This is the only explanation that makes sense.

I will still run a test. I should NOT have to remove the failure routing settings at my provider for what I believe is a 3cx bug.

thanks leon
 
I will still run a test. I should NOT have to remove the failure routing settings at my provider for what I believe is a 3cx bug.
If that is the case and the call ends up in the failure routing of the provider that means that the call is being rejected by the PBX for some reason.
So again check where this fails and what is happening exactly so we are able to better advise.
 
Hi there...I just ran a few tests. I've attached it here my notes are in the file but basically yes it seems to be taking the provider failure route. Also, an extension in the RG I re-added never rang (3cx client) either yet dialing that extension (715) works from 715 on a GS Ip phone. Dialing the RG800 directly worked as expected but 715 client didnt ring either.
 

Attachments

So...is the call taking the Voipms failure route because...

The 3CX trunk is not registered? (which you can confirm in 3CX)

The call is sent to 3CXand "rejected" ?(wireshark, or Voip.ms can confirm)

The call is sent to 3CX and is not answered within a set time, is then sent to alternative destination? (check if there is a "no answer" time limit at Voip.ms)
 
From the attached log file i can see 2 main issues. The first is that there seem to be source identification issues. That means that the PBX cannot match the incoming call to a trunk.

Do you have multiple trunks of the same provider on the PBX?
Do you have the same DID on more than one trunks?
Does the format of the DIDs match the format the provider is sending the number in to the PBX?

The second issue is that from what i can tell you are trying to call the ring Group 800 from extension 715 which is part of the Ring Group. In this scenario extension 715 will not ring.
 
Hi there...yeah I dont know why I am getting those errors. I have three trunks, voip.ms, vitelity and tollfree. currently one did from voip.ms is routed to this pbx. none are routed on vitelity. tollfree is outbound. (and outbound routing is another thread here). This is a very basic configuration.

why won't 715 ring from 800 if I dial 800 from 715? all extensions are set for multiple calls.
this is something I can do on asterisk.

i'm going to try and run a live test with voip.ms since they can't pull older logs.

Leon
 
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