After some tests, I figured out, that it's the enterprise originate which prevents signalling of missed calls. As soon as I use "," instead of ":_:", the signalling of missed calls is fine on the phones.

Question: Is there any reason for this behaviour, or should we consider this as a defect and open a Jira?

Best regards
Peter


On 12/19/17 14:23, Peter Steinbach wrote:
Hello.

I did some further testiumng and figured out, that as soon I use enterprise originate with :_: , I do not see any missed calls. If I use a comma instead, missed calls are shown on the phone. But then as a drawback only one phone rings, if I have more than physical phone registered to one user extension.

Anybody has an idea how to overcome this?

Best rgeards
Peter

On 12/11/17 17:30, Peter Steinbach wrote:
Hello,

we have some Problems with the right signalling of missed calls when calling multiple phones in parallel

Here's the scenario:
Phone no 49 is calling a group with 2170 and 3275 with the following dialstring

<action application="bridge" data="{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59eee5e0-c0bc-0135-7dc9-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/2170@flex.mydomain.de:_:{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59f07af0-c0bc-0135-7dca-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/3275@flex.mydomain.de" />

Destination dialstrings are seperated by ":_:" ("Enterprise Origination"). We use curly brackets instead of "<" as we sometimes have to insert asserted identy tags into the dialstring.

We checked 3 versions of Freeswitch for this
So for the 2 never Freeswitch Versions, here are the logs at hangup
2017-12-11 17:07:46.022450 [DEBUG] sofia.c:7283 Channel sofia/internal/49@flex.mydomain.de:5060 entering state [terminated][487]
2017-12-11 17:07:46.022450 [NOTICE] sofia.c:8474 Hangup sofia/internal/49@flex.mydomain.de:5060 [CS_EXECUTE] [ORIGINATOR_CANCEL]
2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup sofia/internal/2170@94.xx.xxx.xx:42170 [CS_CONSUME_MEDIA] [LOSE_RACE]
2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot create outgoing channel of type [user] cause: [LOSE_RACE]
2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate Resulted in Error Cause: 502 [LOSE_RACE]
2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/2170@94.xx.xxx.xx:42170) Running State Change CS_HANGUP (Cur 3 Tot 22)
2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/2170@94.xx.xxx.xx:42170) Callstate Change RINGING -> HANGUP
2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/2170@94.xx.xxx.xx:42170) State HANGUP
2017-12-11 17:07:46.042331 [DEBUG] mod_sofia.c:449 Channel sofia/internal/2170@94.xx.xxx.xx:42170 hanging up, cause: LOSE_RACE
2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup sofia/internal/3275@94.xx.xxx.xx:43275 [CS_CONSUME_MEDIA] [LOSE_RACE]
2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot create outgoing channel of type [user] cause: [LOSE_RACE]
2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate Resulted in Error Cause: 502 [LOSE_RACE]

Here is the Cancel message for one of the called phones:
U 2017/12/11 17:07:46.045120 144.xx.xxx.xx:5060 -> 94.xx.xxx.xx:42170
CANCEL sip:2170@94.xx.xxx.xx:42170 SIP/2.0.
Via: SIP/2.0/UDP 144.xx.xxx.xx;rport;branch=z9hG4bK07r3ce94cvjpp.
Max-Forwards: 70.
From: "Test" <sip:49@flex.mydomain.de>;tag=mQS1eFDpp1peS.
To: <sip:2170@94.xx.xxx.xx:42170>.
Call-ID: 4238b5a7-5930-1236-b2ab-00505600a1a5.
CSeq: 116168167 CANCEL.
Reason: SIP;cause=200;text="Call completed elsewhere".
Content-Length: 0.

Any hints why this happens, or anyone has this scenario working?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




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-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de