On cube make sure you specify the source address on your dial-peers voice-class sip bind media|control to the correct side. I have seen one way audio when not set. On 5 Aug 2013, at 06:29, Peter <eidevm5@gmail.com> wrote:
I currently have successful two way calls (signalling and media) in the following setup
External Linphone --> Freeswitch --> Freeswitch SBC -> Router -> Kamailio --> Internal Linphone
However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
External Linphone --> Freeswitch --> Freeswitch SBC -> Router -> CME -> CUCM9 --> Cisco handset
The call signalling appears to be working fine and I can successfully initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.
Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
I tried setting adding:
<action application="set" data="disable_rtp_auto_adjust="true" />
to the dialplan on the SBC, but it made no difference.
Any suggestions as to what to check next?
Thanks
Peter
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org