CallManager - SIP Trunk through an ASA. Has anyone got this working?

ccpagel · ‎10-15-2007

Hi all,

I'm trying to create a SIP Trunk to a SIP service provider from a CallManager 4.2 server.

Has anyone got this working?

I get inbound/outbound call setup OK but no media. I've allow all IP traffic inbound from the external SIP server and also port forwarded UDP port 5060 to the CCM server.

I'm assuming that the ASA is clever enough to inspect the SIP signalling traffic and forward the media to the appropriate inside address (MTP or Transcoder depending on what we are doing) however I must ne missing something as this doesn't work.

Thanks,

Chris

Report Inappropriate Content · ‎10-19-2007

I think ASA is blocking RTP stream between the two points

RTP needs UDP 16384 through 32768 open for most VOIP applications, and SIP ports are 5060 UDP/TCP.

http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/c1693/ccmigration_09186a00807217e1.pdf

ccpagel · ‎10-19-2007

On the ASA we have permitted all traffic from the SIP server but it doesn't help.

I think the problem is that the ASA looks at the SIP invite and dynamically tries to open ports and create NAT entries but its not working.

Chris