Dropped Call On New CISCO Router For No Reason!

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Hello Everyone,

Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT
setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected.

Your help is greatly appreciated,

Nick.

3 thoughts on - Dropped Call On New CISCO Router For No Reason!

  • This is a classic symptom of having reinvites and/or direct media enabled on Asterisk or SIP ALG enabled on the router.

    —–Original Message—

  • Hello Eric, I knew this problem all so well however, never knew CISCO sip alg was enabled by default. The following settings got us up and going shortly after the email:

    no ip nat service sip udp port 5060

    ip nat inside source static udp 192.168.2.5 5060 interface Dialer0 5060

    access-list 130 permit udp any any range 8000 65535
    route-map voip-rtp permit 1
    match ip address 130
    ip nat inside source static route-map voip-rtp

    Happy New Year to All,

    Nick from Toronto.