“Cannot complete conference.” When I am trying to make conference with the PSTN Numbers

[Nayeem M.V.]

Hi All,

I have IP Telephony setup with CCME 7.0 and SIP trunk  and I am getting the error as “Cannot complete conference.” When I am trying to make conference with the PSTN Numbers.

The sip dial peer used  G729rb8 codec as per the ISP standard. As per my knowledge that conference feature doesn’t support G729 codec. Is it possible to do transcoding for the calls coming from the SIP dial peer leg , on the same CME itself without putting transcoder  (router as a transcoder ) between the ISP and CME in the customer Local area network .

Please suggest

Nayeem

[iantra123]

Hi,

Have you already tried to make conference with g711?

Can you give the running config

1/ In the DSP farm

2/ In the telephony-Service

[Jaime Valencia]

If your going to use DSPs you could configure a HW CFB or XCODER

Transcoding Support

Transcoding compresses and decompresses voice streams to match  endpoint-device capabilities. Transcoding is required when an incoming  voice stream is digitized and compressed (by means of a codec) to save  bandwidth, and the local device does not support that type of  compression.

Cisco CME 3.2 and later versions support transcoding between G.711 and  G.729 codecs for the following features:

•Ad hoc conferencing—One or more  remote conferencing parties uses G.729.

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrnsc.html

The "Configuring conferencing" has the instructions for HW CFB.

HTH

java

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