Trunked call disconnects

General Help
1

We have two PBX systems. One is at a colo site and is running Elastix PBX. It has a SIP trunk. The second is a FreePBX system located at an office location and has all the DIDs, extensions, ring groups, etc. on it.

For technical reasons (which ar enot currently relevent), we need to have the SIP calls coming in to the Elastix PBX at the colo and trunked over to the FreePBX system. This works no problem (I have an IAX2 trunk conencting the two systems)

We are trying to replace the Elasstix systemwith another FreePBX system. When testing, I am able to swap the Elastix PBX with the replacement FreePBX system we have (not the one at the office location, but one at our colo), and have the SIP calls come through. The IAX2 trunk is conencted to the office FreePBX sytem, and calls come in to the colo PBX and through to the office PBX.

The problem, however, is that while the IVR is playing, the call disconnects. There is no problem when the Elastix PBX is swapped back in. The error logs on the colo FreePBX system (the replacement for Elastix) when this happens look like this:

(identifying information changed, but the error messages themselves are not)

    -- Called SIP/colo-peer/7789451068
    -- SIP/colo-peer-00000006 answered SIP/SIPPROVIDER-00000005
[2014-11-23 09:27:41] WARNING[28797]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 486284637 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2014-11-23 09:27:41] WARNING[28797]: chan_sip.c:4053 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (ext-trunk, tdial, 9) exited non-zero on 'SIP/SIPPROVIDER-00000005'

A search on the error indicates that this is usually due to network or firewall issues, but the only thing that changes network-wise is the swapping of the Elastix PBX for the new FreePBX; the IPs reain the same (I change the IP of the Elastix PBX and give the new FreePBX it’s IP). In all cases, the PBX systems are behind a NAT firewall.

Is there maybe something on a FreePBX system that I need to do that might be affecting this? Maybe something that it doens;t like about being behind a NAT firewall?

Thanks for your help and advise.

Regards,

Alan

2

Check Asterisk SIP settings and make sure you have the external and internal IPs configured correctly.

3

Sorry, I did not think to mention that the “External IP” in “Asterisk SIP Settings” is already set to the IP the FreePBX (and Elastix box) NAT as :frowning: