[asterisk-users] Cisco 18XX series router - one way audio

[Ishfaq Malik]

Hi

Has anyone experienced any issues with calls through asterisk server
with a netted phone connected to a Cisco 18XX series router.

I'm experiencing one way audio when the caller calls from the phone
connected to the asterisk server to the outside world (via a SIP
provider). It's the audio in to the caller that is failing.

We're using asterisk 1.8.7.0

When looking at the logs I'm seeing RTP transmission errors as the RTP
is being router to port 0 on the correct IP address. If I do a SIP trace
I can see that all SIP traffice is being router to the correct port on
the IP address.

I just can't work out where this port 0 is coming from and have a
suspicion that the router is doing something funny.

Has anyone experienced anything like this before?


Thanks in Advance

Ish


-- 
Ishfaq Malik <ish at pack-net.co.uk>
Department: VOIP Support
Company: Packnet Limited
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