12-18-2008 01:57 PM - edited 03-15-2019 03:09 PM
I have a user that has an 871 ISR at hoe with a tunnel bakc to an asa. The user can make call back to the office but cannot hear anything after the call connects. Running 12.4 code and VPN tunnel is stable. Has anyone deployed this option befire that has had similar problems? Thanks in advance.
12-19-2008 08:57 AM
Hello,
I have and in my instance the remote phone IP addresses have access to the CUCM but they did not have access to the HQ phone subnet (RTP flows direct from phone to phone).
hth
Jesse
12-19-2008 09:04 AM
Thanks Jesse. I believe the RTP is the problem here. The phone knows how to get to the users in headquarters but the phone at the HQ don't know how to get back to the remote location for the RTP stream. There is some natting going on on the ASA and I'm not sure where the problem lies. If I attach a config, can you take a look?? This is an installation that was done about a year ago.
12-19-2008 09:25 AM
Try this:
Get a call up with no audio.
Go to the website of the IP address of each phone, click on Stream 1.
Each phone should tell you the IP address it is trying to get to.
Next, go to the default gateway of each of the phones and try pinging the address each of the phones is trying to send audio to. From there, you can troubleshoot from a VPN or ASA level.
If you can ping, that means it's being filtered on the port/protocol. Check your ACLs and firewall settings.
If you're using CME, you may find it helpful to put 'mtp' under the ephone configuration of the remote phone.
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