Conference call inside extentions OK, Outside drops call

[velocityiq]

When using the conference room feature, any user calling in from outside the system (I.E. SIP trunk) gets the prompt "enter conference room ID" and upon pressing any digits, the call gets disconected . We have Broadvoice SIP trunks and BroadVox sip trunks. From inside, at any extention all works well. 2nd interesting symptom is that when the call is dropped, 3CX reports the SIP trunk as not busy (Green) and the log file shows "call was terminated" BUT the calling party (Like a cell phone) shows still connected.
System has been running for over 7 months with no issues and calls are clear and work inside / outside with no problems. ONly when using Conference rooms does this happen.


10:56:59.368 [CM503008]: Call(40): Call is terminated

10:56:59.352 [CM503008]: Call(40): Call is terminated

10:56:58.650 [CM503007]: Call(40): Device joined: sip:[email protected]:40300;rinstance=871906d6f111a62e

10:56:58.650 [MS210004] C:40.1:Offer provided. Connection(proxy mode): 67.78.**.**:****(9007)

10:56:58.634 [MS210000] C:40.3:Offer received. RTP connection: 127.0.0.1:40352(40353)

10:56:58.634 Remote SDP is set for legC:40.3

10:56:58.634 [CM505001]: Ext.702: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Conference place;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Conference Place] Transport: [sip:127.0.0.1:5060]

10:56:58.634 [CM503002]: Call(40): Alerting sip:[email protected]:40300;rinstance=871906d6f111a62e

10:56:58.134 [CM503024]: Call(40): Calling Ext:Ext.702@[Dev:sip:[email protected]:40300;rinstance=871906d6f111a62e]

10:56:58.103 [CM503004]: Call(40): Route 1: Ext:Ext.702@[Dev:sip:[email protected]:40300;rinstance=871906d6f111a62e]

10:56:58.103 [CM503010]: Making route(s) to <sip:[email protected]:5060>

10:56:58.088 Refer: from="CNF701:"<sip:[email protected]:5060>;tag=ef3e7c68; to="DAVID WAGNER"<sip:727422****@127.0.0.1:5060>;tag=79631a44; RefTo=<sip:[email protected]:5060>

10:56:48.892 Session 482 of leg C:40.1 is confirmed

10:56:48.736 [CM503007]: Call(40): Device joined: sip:[email protected]:40300;rinstance=99e9de9f62c259b5

10:56:48.736 [CM503007]: Call(40): Device joined: sip:727422****@209.249.3.59:5060

10:56:48.736 [MS210005] C:40.1:Answer provided. Connection(proxy mode):67.78.**.**:****(9007)

10:56:48.721 [MS210001] C:40.2:Answer received. RTP connection: 127.0.0.1:40350(40351)

10:56:48.721 Remote SDP is set for legC:40.2

10:56:48.705 [CM505001]: Ext.701: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Conference place;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Conference Place] Transport: [sip:127.0.0.1:5060]

10:56:48.705 [CM503002]: Call(40): Alerting sip:[email protected]:40300;rinstance=99e9de9f62c259b5

10:56:48.221 [CM503024]: Call(40): Calling Ext:Ext.701@[Dev:sip:[email protected]:40300;rinstance=99e9de9f62c259b5]

10:56:48.221 [MS210004] C:40.2:Offer provided. Connection(proxy mode): 127.0.0.1:7048(7049)

10:56:48.205 [CM503004]: Call(40): Route 1: Ext:Ext.701@[Dev:sip:[email protected]:40300;rinstance=99e9de9f62c259b5]

10:56:48.205 [CM503010]: Making route(s) to <sip:[email protected].**.**:****>

10:56:48.190 [MS210000] C:40.1:Offer received. RTP connection: 209.249.3.60:22100(22101)

10:56:48.190 Remote SDP is set for legC:40.1

10:56:48.190 [CM505003]: Provider:[broadvox-1] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhoneSystem 7.1.7060.0] Transport: [sip:192.168.**.**:****]

10:56:48.158 [CM503001]: Call(40): Incoming call from 727422****@(Ln.10003@broadvox-1) to <sip:[email protected].**.**:****>

10:56:47.971 [CM503012]: Inbound office hours rule (unnamed) for 10003 forwards to DN:701

10:56:47.971 Looking for inbound target: called=727475****; caller=727422****

10:56:47.924 [CM500002]: Info on incoming INVITE:

 

[SY]

Please specify version of 3CX PBX and type of your license.
also specify settings "Supports reinvites", "supports replaces" and "PBX delivers audio" which are applied for "broadvox-1" VoIP provider(line).

Thanks

 

[velocityiq]

3CX version 7 3CXPSSBNFR
Supports re-invite = Yes
Supports replace = yes
PBX delivers audio = no

 

[SY]

3CX version 7 3CXPSSBNFR
Supports re-invite = Yes
Supports replace = yes
PBX delivers audio = no

What is the reason to ignore recommendation? (this configuration is, definitely, not recommended)
other questions:
1. Can you put call (received from this broadvox line) on hold?
2. can you transfer call to external number using broadvox line?
3. can you transfer call (received from this broadvox line) to another extension or using IVR?

Thanks

 

[velocityiq]

That is the default that was setup when I established the line. I have not changed them. Yes you can transfer calls and yes you can forward to outside line. I don't know if IVR works, have not used that. I will adjust those two settings and test / post response.

 

[SY]

That is the default that was setup when I established the line. I have not changed them. Yes you can transfer calls and yes you can forward to outside line. I don't know if IVR works, have not used that. I will adjust those two settings and test / post response.

Hmm...
I've checked both templates for broadvox ("Broadvox GoAnywhere" and "Broadvox SIP Trunk") and see that default settings are:
Supports re-invites = no
Supports replaces = no
PBX delivers audio = yes

Question:
why do we have the different default values in our templates? (I've used v8 to compare them and slightly sure that they are the same in v7.1)

Thanks

P.S. Please provide the logs in verbose mode for the unsuccessful call to conference place (most interesting is 3CXPhoneSystem.trace.log and 3CXMediaServer.trace.log). It is really difficult to understand what is going wrong without information. Thanks once again.

 

[cyborg]

I have a similiar problem. When using conferencing on the inside all works ok, when coming in on a Generic SIP trunk virtual extension 10000 thru X, it seems to not allow me to join a confrence in progress. It alway sets up a new confrence on the next 700 series extension. Running paid for version 9.0.15781.949

All other functionality of the generic SIP gateway is working fine. I can forward to a particular extension or to digital assistant just fine. Conference mode is the only issue at this time. PBX Delivers audio is the only box checked in the Gateway configuration. Any advise on whether Re-Invite or replaces should be checked ?

Thanks Much,

Cyborg