Cisco Conference Now

  • A maximum of 100 simultaneous Conference Now and Meet-Me conference are supported per CUCM node
  • The maximum number of conference parties is controlled by the existing CallManager service parameter Maximum Meet-Me Conference Unicast
  • Once you are placed into the conference, you are no connected to the IVR CUCM - but connected to configured CUCM Conference Resource
  • The Conference Resource selected by CUCM (confirmed from testing) is determined by device MRGL of the Conference Host , i.e. the Conference Host device (e.g. their phone's MRGL or if opening externally via the specific gateway's MRGL)
  • Once the Host opens the bridge the bridge stays open until all callers leave the bridge (even if host leaves the bridge)
  • DTMF Digits - The IVR supports only Out-Of-Band (OOB) DTMF digit collection method. If there is a DTMF capability mismatch between the calling device and the IVR, an MTP will be allocated.
  • Codecs - The IVR only supports codec G.711 (a-law and mu-law), G.729, and Wide Band 256k. If there is a codec mismatch between the alling device and the IVR, a transcoder will be allocated

For SIP Calls coming in a gateways - you must include sip-kpml as a SIP DTMF relay option, otherwise an MTP resource will be required for the duration of that call.

The Below voice dial-peer dtmf-relay configuration will both RFC 2833 (rtp-nte) and 'Out-of-band (OOB)' sip-kpml.

dtmf-relay rtp-nte sip-kpml

Note: there is no difference in the SIP Invite and SDP if you change the order of above. i.e. there is no technical way to inform the other SIP endpoint you have a preference for KMPL over RFC2833 (NTE). But Cisco phones do 'prefer' NTE mode - when both are offered (but they support both).

Annoucment Identifier Annoucement Text Notes
ConferenceNowGreeting Welcome. To Join a Conference Now, please enter a meeting number, following by hash. Include a call recording warning if call is recorded, e.g. 'Welcome to Purplepi Conferencing. Your Call May be recorded for training and quality purposes. Please enter a meeting number, following by hash.'
ConferenceNowNumberInvalid The meeting number that you entered is invalid, please re-enter the meeting number
ConferenceNowEnterPIN If you are the meeting host, please enter your PIN, otherwise press hash to continue.
ConferenceNowInvalidPIN The PIN that you entered was invalid, please re-enter your PIN
ConferenceNowEnterAccessCode Please enter the attendee access code, followed by hash.
ConferenceNowAccessCodeInvalid The meeting access code that you entered is invalid, please re-enter the meeting access code
ConferenceNowNumberFailed The meeting number that you entered is invalid, please ask the meeting host for the correct meeting number. Goodbye. after 5 failed attempts
ConferenceNowAccessCodeFailed The meeting access code that you entered is invalid, please ask the host for the correct access code. Goodbye after 5 failed attempts
ConferenceNowFailedPIN The PIN that you entered is invalid, please ask the system administrator for the correct meeting number and PIN. Goodbye. after 5 failed attempts
ConferenceNowCFBFailed <not sure of exact wording!> Note this prompt for NOT play if the System Parameter “IVR” Call Count is reached. Call is just rejected!
i.e. even though the documents state 'Plays when the conference bridge capacity limit is exceeded while
initiating Conference Now'
I assume it will be played if more than 100 concurrent conference are reached on a specific node.

Note Use of an MTP.Why?

Once user joined the conference, if the bridge is not opened, the user hears MoH and the MTP is removed from the call.

#show dspfarm all

...
...

Dspfarm Profile Configuration

 Profile ID = 13, Service = MTP, Resource ID = 5
 Profile Service Mode : Non Secure
 Profile Admin State : UP
 Profile Operation State : ACTIVE
 Application : SCCP   Status : ASSOCIATED
 Resource Provider : FLEX_DSPRM   Status : UP
 Total Number of Resources Configured : 3
 Total Number of Resources Available : 2
 Total Number of Resources Out of Service : 0
 Total Number of Resources Active : 1
 Hardware Configured Resources : 3
 Hardware Resources Out of Service: 0
 Software Configured Resources : 0

 Number of Hardware Resources Active : 0
 Number of Software Resources Active : 1
 Codec Configuration: num_of_codecs:2
 Codec : pass-through, Maximum Packetization Period : 0
 Codec : g711ulaw, Maximum Packetization Period : 30


#show call active voice compact
 <callID>  A/O FAX T<sec> Codec       type        Peer Address       IP R<ip>:<udp>        VRF
Total call-legs: 4
        89 ANS     T5     g711alaw    TELE        P768978692
        90 ORG     T5     g711alaw    VOIP        P75912             <This Gateway IP address>:16406
        91 ORG     T5     pass-throug VOIP        P                  <This Gateway IP address>:16404
        92 ORG     T5     pass-throug VOIP        P                  <CUCM Conference Now IVR server>:24856
#show dspfarm all
Dspfarm Profile Configuration

 Profile ID = 16, Service = TRANSCODING, Resource ID = 1
 Profile Service Mode : Non Secure
 Profile Admin State : UP
 Profile Operation State : ACTIVE
 Application : SCCP   Status : ASSOCIATED
 Resource Provider : FLEX_DSPRM   Status : UP
 Total Number of Resources Configured : 3
 Total Number of Resources Available : 2
 Total Number of Resources Out of Service : 0
 Total Number of Resources Active : 1
 Codec Configuration: num_of_codecs:5
 Codec : pass-through, Maximum Packetization Period : 0
 Codec : g711ulaw, Maximum Packetization Period : 30
 Codec : g711alaw, Maximum Packetization Period : 30
 Codec : g729ar8, Maximum Packetization Period : 60
 Codec : g729abr8, Maximum Packetization Period : 60

...
...

 Number of Hardware Resources Active : 0
 Number of Software Resources Active : 0
 Codec Configuration: num_of_codecs:2
 Codec : pass-through, Maximum Packetization Period : 0
 Codec : g711ulaw, Maximum Packetization Period : 30

#show call active voice compact
 <callID>  A/O FAX T<sec> Codec       type        Peer Address       IP R<ip>:<udp>        VRF
Total call-legs: 4
        89 ANS     T19    g729r8      TELE        P768978692
        90 ORG     T19    g729r8      VOIP        P75912             <This Gateway IP Address>:16410
        96 ORG     T5     pass-throug VOIP        P                  <This Gateway IP Address>:16404
        97 ORG     T5     pass-throug VOIP        P                  <Branch Gateway IP Address (Conference Resource)>:16700

Note: Host is located at the branch for this call flow

Branch#show dspfarm all
Dspfarm Profile Configuration

 Profile ID = 1, Service = CONFERENCING, Resource ID = 1
 Profile Description :
 Profile Service Mode : Non Secure
 Profile Admin State : UP
 Profile Operation State : ACTIVE
 Application : SCCP   Status : ASSOCIATED
 Resource Provider : FLEX_DSPRM   Status : UP
 Total Number of Resources Configured : 2
 Total Number of Resources Available : 1
 Total Number of Resources Out of Service : 0
 Total Number of Resources Active : 1
 Maximum conference participants : 8
 Codec Configuration: num_of_codecs:6
 Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required
 Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required
 Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required
 Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required
 Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required
 Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required


SLOT   DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0      1   40.2.10  UP     1    USED  conf    1      75        106836    106827
0      1   40.2.10  UP     1    USED  conf    1      82        50745     50724
0      1   40.2.10  UP     N/A  FREE  conf    1      -         -         -

Total number of DSPFARM DSP channel(s) 2


Branch#show call active voice compact
 <callID>  A/O FAX T<sec> Codec       type        Peer Address       IP R<ip>:<udp>
Total call-legs: 2
      2202 ORG     T2161  g729r8      VOIP        P      <IP address of Host Phone>:16852
      2210 ORG     T52    g729r8      VOIP        P      <IP address of Gateway from external Attendee>:16426

using the show sip-ua calls* on the gateway you can confirm the Dtmf-relay Negotiated, e.g. in below example - it is kpml**. Note: the 'show sip-ua calls dtmf-relay sip-kpml' does not seem to work. As it shows '0' calls even if a call is using KPML as per below.

GATEWAY#show sip-ua calls
Total SIP call legs:1, User Agent Client:1, User Agent Server:0
SIP UAC CALL INFO
Call 1
SIP Call ID                : [email protected]
   State of the call       : STATE_ACTIVE (7)
   Substate of the call    : SUBSTATE_NONE (0)
   Calling Number          : +353768978692
   Called Number           : 75912
   Called URI              : sip:[email protected]
   Bit Flags               : 0xC04018 0x90800100 0x80
   CC Call ID              : 205675
   Local UUID              : 6a58d8d700105000a000006cbca9ea0a
   Remote UUID             : a7c4ddf6abb85f1b94df1c05a987d304
   Source IP Address (Sig ): 10.10.20.30
   Destn SIP Req Addr:Port : [10.10.10.1]:5060
   Destn SIP Resp Addr:Port: [10.10.10.1]:5060
   Destination Name        : 10.10.10.1
   Number of Media Streams : 1
   Number of Active Streams: 1
   RTP Fork Object         : 0x0
   Media Mode              : flow-through
   Media Stream 1
     State of the stream      : STREAM_ACTIVE
     Stream Call ID           : 205675
     Stream Type              : voice-only (0)
     Stream Media Addr Type   : 1
     Negotiated Codec         : g729r8 (20 bytes)
     Codec Payload Type       : 18
     Negotiated Dtmf-relay    : sip-kpml
     Dtmf-relay Payload Type  : 0
     QoS ID                   : -1
     Local QoS Strength       : BestEffort
     Negotiated QoS Strength  : BestEffort
     Negotiated QoS Direction : None
     Local QoS Status         : None
     Media Source IP Addr:Port: [10.10.20.30]:17708
     Media Dest IP Addr:Port  : [10.10.30.99]:26994
   Mid-Call Re-Assocation Count: 0
   SRTP-RTP Re-Assocation DSP Query Count: 0


Options-Ping    ENABLED:NO    ACTIVE:NO
   Number of SIP User Agent Client(UAC) calls: 1

SIP UAS CALL INFO
   Number of SIP User Agent Server(UAS) calls: 0
  • vendors/cisco/uc/cucm/conferencenow.txt
  • Last modified: 2019/08/28 11:37
  • by gerardorourke