Recent content by Adrian Fretwell

DjangoPBX version 0.9.6 is out. It includes the following new features and bug fixes: New Features added API clickdial function for Admin users. added option to disbale error file in play/rec_and_get_digits. added download option for voicemail greeting in API. added domain_id filter, where...

That issue is fixed in the master branch. I will get a new release issued this afternoon.

Sorry I can't be of more help, I have never had to do this because we don't have any switches behind NAT. You could try export bypass_media=true. It will probably help you to get some packet captures and look at what IP addresses and ports are being specified in the SIP SDP bodies.

Your problem is indeed data tromboning. Your best solution will be to instruct Freeswitch to bypass the media for the forwarded call. It can be done on the fly and there is a call forwarding example in the following link. If you record calls, recording will not work with media bypassed...

DjangoPBX is capable of being run in such a way, using it's clustered configuration, but this is not a turnkey solution and requires a good level of technical ability. https://www.djangopbx.com/documentation/cluster/typical_cluster_diagram.html

Just be aware that using the nftables rule set above, will mean that non of the counters or sets that DjangoPBX interacts with will be available. Any firewall related activity will fail. For example, when someone requests too many URLs that do not exist, their IP address is automatically put...

OK, it is an interesting one, I will give it some more thought. One thing I am curious about is, as it is WebSocket transport, which I believe is a "sub protocol" of TCP, why are you seeing these brief disconnections, normally TCP would protect you from such transient errors. But as you point...

You may be trying to fix the symptom instead of the Illness. I would be more concerned about why the agents lose the connection in the first place.

I was going to suggest raising this with the Singalwire Freeswitch Forum, but I see you have already done that. https://support.signalwire.com/portal/en/community/topic/freeswitch-481-response-to-bye I can't think of anything else to help right now, but I will post if I do have any ideas. If...

I have seen this happen where a SIP ALG or helper in the router/firewall corrupts the Call-ID. Some endpoints use their IP address as part of the Call-ID, if this IP address is an RFC1918 address many ALGs and "helpers" will try to fix it by replacing it with what it believes is the correct...

Have a look at toll_allow for outbound.

I would try to get packet captures both on your PBX box and at the customers premises. Reading them may give you some clue.

I agree, looking at the FreeSWITCH source code it what I have often had to do to find out about undocumented configuration, a good example was mod_amqp. As for AI, It is my personal opinion that it may be a useful learning tool although it is not always correct or accurate. I certainly would...

As you are probably aware FreeSWITCH will make the CURL call to your API and provide lots of variables that help you decide what FreeSWITCH is asking for. For example it will give to the number dialed and the domain name. Probably the most important POST variable that you will read is...

When I last played with fusion there was a global dialplan record called user_exists. This did many things but crucially it set a channel variable called ${from_user_exists}. All you need to to is check if this is true in your outbound route. (user_exists makes use of the sip_from_user...