I've been experimenting with asteriskNow (freePBX/asterisk 11) as a sip server with DDNS. Everything seems to be working fine using externhost and such; calls to remote extensions to and from asterisk are going through. There is also a grandstream ATA (HT503) with an FXO line also setup in this scenario. Branched to the FXO line is a physical extension from a plain old Panasonic PBX on which any analog telephone extension (no voip magic here) usually dials 82 for a CO line. From within the voip network, i dial the asterisk FXO extension or SIP ID through a voip phone, get the panasonic dial tone, dial 82, and reach the CO dial tone.
When i try the same thing remotely over DDNS, i dont reach the CO dial tone. What's puzzling me is that i can reach the panasonic extensions after dialing the FXO sip id and reaching that first dial tone, so what could be the deal with that second dial tone only disappearing with ddns?


For the sake of simplicity, i decided to use the grandstream ATA directly with a CO line thus removing the panasonic pbx from the equation. from my remote extension, dialing the extension of the FXO causes a temporary disconnection of the server-side internet. There is nothing notable in the sip debug of this call except a hangup due to lack of rtp activity. A call from a local extension to the fxo is still working fine.

Any other call from remote ddns extension to local or remote (non-fxo) extension gets through without a glitch. what should i be looking at? the ATA itself? or the modem on the server side? or the asterisk configuration?


That mean you have no port forwarding on external router, SIPALG enabled on router or not setuped NAT correctly.

For more info see sip debug and study this article


  • I tried disabling sipalg and it still didnt work. just to be sure we're on the same page: i can reach the fxo dial tone from my remote voip phone, i can dial an internal analog extension successfully (all over dyndns). the only thing i can't do is reach the second CO dial tone from the fxo's first dial tone. if port forwarding is an issue, shouldnt i be unable to place the first call to the remote sip account registered to the ATA's FXO to begin with? – 3a2roub Nov 28 '14 at 13:17
  • Sorry,debug is outside this site topic.Check sip trace by tcpdump/wireshark – arheops Nov 28 '14 at 14:25

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