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Call dropping issue from a conference bridge

NSG Manager
Level 1
Level 1

Hi

Currently we are facing issues of call dropping from conference bridge, We do have Cisco meeting place express 2.0 for conferencing and CiscoCallManager 5.1 as IPBX ,Call dropping occurs only on a multiparty voice conference using cisco meeting place express.

Appreciate an advice regarding how to trouble shoot the issue.

Thanks in Advance

Best Regards

Vijay

3 Replies 3

drolemc
Level 6
Level 6

In order to workaround this issue, complete one of these:

Disable Transcoding (or) Make sure that the Transcoding and Voice Termination are on different DSPs. In order to do this, configure the maximum number of transcoding sessions to a value such that it requires a multiple of 240 DSP credits.

Hi Drolemc,

We have two profiles configured in our CMM ,please find the running config below.

mediacard 4

resource-pool xcode dsps 1

resource-pool cfb dsps 3

!

!

sccp local GigabitEthernet1/0

sccp ccm 172.16.0.6 identifier 1

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate profile 2 register C07001AA1483C0D ( This is registerd in call Manager)

associate profile 1 register M08001AA1483C0C (This is registered in call manger)

!

dspfarm

!

dspfarm profile 1 transcode ========> Used for External calls to translate

rtp timeout 7200

codec g711ulaw packetization-period 30

codec g711alaw packetization-period 30

codec g729r8 packetization-period 30

codec g729ar8 packetization-period 30

codec g723r63 packetization-period 30

codec g723r53 packetization-period 30

associate resource-pool xcode

!

dspfarm profile 2 conference adhoc ========> Used for Meet me and Adhoc conferencing only

rtp timeout 7200

codec g711ulaw packetization-period 30

codec g711alaw packetization-period 30

codec g729r8 packetization-period 30

codec g729ar8 packetization-period 30

codec g723r63 packetization-period 30

codec g723r53 packetization-period 30

associate resource-pool cfb

!

call manager 5.0 and Meeting place 2.0 are on g711 hence persume that it do not use any of the exixting resource DSP resources

PSTN=====> 2821 Router Vgw ======> Call Manger 5.0 =======> Meeting place 2.0

We have also integrated the PABX with gateway router using a different PRI.

Voice termination is on the voice gateway router and using the resource as shown below (from the voice gateway)

PVDM Slot 0:

64-channel (G.711) Voice/Fax PVDMII DSP SIMM PVDM daughter card

The logs from the gateway router shows the following error:

989726: Aug 19 14:56:11.114: H245 MSC OUTGOING ENCODE BUFFER::= 23800001

989727: Aug 19 14:56:11.114:

989728: Aug 19 14:56:11.118: %ISDN-6-DISCONNECT: Interface Serial0/0/0:20 disconnected from 8067072256 , call lasted 1249 seconds

989729: Aug 19 14:56:11.118: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x803D

Cause i = 0x80A9 - Temporary failure

Please do advice as to the changes I got to make in the configurations for a work around

Thanks in Advance

Vijay

Vijay,

Do you have sufficient bandwidth for conference calls. It drops the calls after how many people joined the conference?.. Can you prioritise your voice traffic with sufficeint bw and set the QoS.

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