Reception Hunt Pilot transfer to MeetMe Number disconnects

[Chadster766]

Reception Hunt Pilot transfer to MeetMe Number disconnects.

Call Flow:

FXO Line => CTI 6000 => CUC Call Handle 6000 => Digit 0 => Call Handler 4505 => Transfer Rule (STD) => Hunt Pilot 4505 => Reception Answers DN 4000-4005 (Longest Idle Time) => Transfer to MeetMe DN 5550

[Chris Deren]

Is MeetMe already started? Remember to join meetMe bridges you first need to open it by pressing the Meetme softkey on a phone.

[Chadster766]

Yes testing was done with the MeetMe conference started before reception transfer to 5550.

[Chris Deren · Accepted solution]

Can the receptionist start the meete? This would confirm proper partition/CSS issue.

What codec is the conference bridge using? Might be codec negotiation issue, reviewing CM trace files would reveal this or any other cause.

[Chadster766]

The receptionist can create the MeetMe but as soon as she transfers one of the 4000-4005 DN to the MeetMe DN they get disconnected.

I did more testing and all reception DN's can create the MeetMe conference.

[Jaime Valencia]

I'd try a simpler test, call the reception directly, can she transfer the call to the meet-me DN?

What protocol are you using on the GW?

And just to be clear, someone else, not the receptionist, is the one who starts the meet-me and waits for the other participants to join??

[Chadster766]

I'm using the Software CFB. Yes everyone can create and join a MeetMe conference the way I have it configure at the moment.

Just incoming FXO calls through the Call Handlers are being disconnected once transferred to the MeetMe.

[Jaime Valencia]

OK, what protocol are you using on the GW with that FXO?

[Chadster766]

voice service voip
 ip address trusted list
  ipv4 172.27.199.11
  ipv4 172.27.199.14
  ipv4 172.27.199.15
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 no supplementary-service sip handle-replaces
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no fax-relay sg3-to-g3
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface BVI1
  bind media source-interface BVI1
  asymmetric payload full
  no update-callerid
!
!
voice class uri ucm sip
 host 172.27.199.11
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
 codec preference 3 g711alaw
 codec preference 4 ilbc
!

I'm not sure this is below section is much help since I'm using the software CFB:

mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp ccm 172.27.199.11 identifier 1 priority 1 version 7.0
!
sccp ccm group 1
 bind interface BVI1
 associate ccm 1 priority 1
 associate profile 1 register CFB1ARBORG
!
!
ccm-manager sccp local BVI1
!
dspfarm profile 1 conference
 description Arborg Conference Bridges
 codec g729br8
 codec g729r8
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec g722-64
 codec ilbc
 maximum sessions 3
 associate application SCCP
!

[Chadster766]

It was codec mismatch between the Conference Bridge Device Pool and Gateway codex.

Thanks everyone for the support.