Remote Extensions Call Drops After 20 Seconds.

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Asterisk Users 14 Comments

Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds.

I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f

Thank you!

14 thoughts on - Remote Extensions Call Drops After 20 Seconds.

  • Calls dropping after 20 seconds is often directmedia enabled when it should not be enabled or RTP keepalives enabled when they should not be enabled. Dropping around 20 mins is often Session Timers being enabled when they don’t work for the specific environment.

    —–Original Message—

  • here’s a checklist…

    First, RTP port range not port forwarded correctly on the NAT router (check rtp.conf).

    Then, on sip.conf:

    externip not correctly setup (it should be the public IP of the NAT
    router)?
    nat setting not enabled for any outbound trunk and the extensions (nat=yes)
    ?
    localnet not properly setup (to include subnets of local, un-nat’d extensions) ?
    canreinvite not disabled for any outbound trunk and for the extensions?

    rgds

  • Rodrigo, thanks for reply.

    1- RTP ports is forwarded correctly on the NAT router.
    2- externip is my public ip.
    3- All my extensions have nat=yes by default.
    4- localnet is setup.
    5- canreinvite is disabled.

    It could be a codec mistake?

  • What version of Asterisk? directmedia=no should be used in versions of Asterisk 1.8 and later.

    —–Original Message—

  • When the call is setup I see your Asterisk retransmitting the “SIP/2.0
    200 OK” packet many times and getting no response. The other end needs to receive the packet and generate an “ACK”. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem.

  • See sip.conf.sample in the Asterisk tarball for documentation of valid settings.

    —–Original Message—

  • Try ulaw instead of g729, set directmedia=no

    I see you are using FreePBX. I cannot help further.

    —–Original Message—

  • Thanks Steve.

    I think my problem is NAT. I’m using iptables, but I don’t sure if I’m doing right steps.

    In the principal router I’ve forwarded the ports, but in my firewall
    (iptables on PBX server) I’m not sure. 201.237.180.154 is my remote place.

    #El NAT para el 5060 y el 10000-30000 (rtp)
    iptables -t nat -A PREROUTING –dst 201.237.180.154 –proto udp –dport
    5060 -j DNAT –to 192.168.1.180
    iptables -t nat -A PREROUTING –dst 201.237.180.154 –proto udp –dport
    10000:30000 -j DNAT –to 192.168.1.180
    iptables -t nat -A PREROUTING –dst 201.237.180.158 –proto udp –dport
    5060 -j DNAT –to 192.168.1.180
    iptables -t nat -A PREROUTING –dst 201.237.180.158 –proto udp –dport
    10000:30000 -j DNAT –to 192.168.1.180
    iptables -t nat -A POSTROUTING –proto udp –src 192.168.1.180 -j MASQUERADE

    iptables -t filter -A FORWARD –proto udp –dport 5060 -j ACCEPT
    iptables -t filter -A FORWARD –proto udp –dport 10000:30000 -j ACCEPT

    Can somebody help me to configure my NAT on iptables ? Maybe an example. Thank you again.

  • Guys, but ALL MY INCOMING CALLS (in remote extensions) WORKS FINE. Should be a NAT issue?