Conference CME between pstn and remote site

[juan.hinojosa89]

Hi, we are implementing a CME, and we are making some tests and all is working fine except one thing. When a sip phone registered to cme make a call to a remote site (the communication between the remote site is h323), the call is established without problems, but if you add a phone from pstn to a conference, the sip phone registered to cme and the pstn phone have audio between them, but the phone from the remote site does not. He cant hear us and we cant hear him. The conference remains established between three phones, but the remote site have no audio at all.

Maybe the  "media flow-through" command into the voice service voip configuration can help. What do you think?

Thank you

[Mohammed al Baqari]

Audio silence are mostly because of routing broken or firewall rules
filtering ports. Who is initiating the conference? Based on that check the
invoked conference bridge. The last check is to find the reachability
between the remote CME and conference bridge IP.

[juan.hinojosa89]

Hi Mohammed,

The conference bridge is the local CME, and there is no problem when we make a call to the remote site. So the remote phone system (it is a PBX) should be reacheable and be able to reach the local CME (I think). Another thing that we are seeing is that when the local phone put the remote phone on hold, the remote phone does not hear the MoH, but when the local phone exit the hold, there is audio again. The MoH server is the local CME also. Thank you!!

[R0g22 · Accepted solution]

SIP CME calls over H.323 trunk has never been a supported design and I doubt that it will ever be supported since H.323 is basically dead now -
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCub00715/?reffering_site=dumpcr

Recommendation would be to move your H.323 trunks to SIP ASAP.

[Mohammed al Baqari]

Can you post your config?

[juan.hinojosa89]

Yes, actually it is already attached!