Cisco Conference Now
Key Aspects of 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
MTP resources
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).
Audio Prompts
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. |
Head Office Gateway
Caller has Calling into Conference Now
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
Caller has joined the conference
#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
Branch Gateway
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
DTMF Debug
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