Search results

I cannot provide specific help like this, it would take time away from my paying customers. To stay connected you will probably need a conference room. My guess is that you would need to create a conference room (maybe on the fly), and then from a specific dial plan condition run some lua to...

In menu->Advanced->Group Manger, you need to have the zoiper permission enabled for the user trying to access it:

I'm sorry, I'm not sure what you mean by "subscribe to help posts". I just click on the "Whats new" tab, and any post that hasn't been answered, I try to provide some help. I normally leave unanswered posts a day before responding unless I absolutely know what the answer is. There are a lot of...

John, the command to reload ACL from cron would be fs_cli -x "reloadacl" but I don't understand why this solves the issue. When the server reboots and FreeSWITCH starts up it would load it's ACL then. Unless maybe there is a timing/dependency issue with PostgreSQL not being ready? Do you...

John, I have had a good look back through my notes and have not come up with anything that will definitely help you, but I'm pretty sure this does all revolve around early media. When a phone receives a 180 ringing, without SDP, it will generate ringing tone (ringback) locally. When a phone...

SIP messages for situations like DND are all replies and as such do not go through the dialplan, this is why FusionPBX uses an lua script to pick up the SIP Code, for example SIP:486 for USER_BUSY, and take the appropriate action. There is no reason why you cannot write your own failure...

Yes, that makes perfect sense. 170.0.0.128/29 is your network which will have usable IPs from 170.0.0.129 to 170.0.0.134. So the first IP is used as your gateway. This is a "virtual" IP for the Virtual Router Redundancy Protocol (VRRP) and I assume that 130 and 131 are the real IPs used...

These IP addresses don't quite make sense to me. It looks like they are giving you a /29 subnet. This would normally give you 6 useable IP addresses, one of which must be the gateway, leaving 5 for your machines. It is normal practice to have the gateway within the subnet. 170.0.0.1 is not...

We had something similar to this a few years ago, I can't remember all the detail right now but I'm sure it was to do with an endpoint not responding correctly to Early Media (183 Session Progress). 180 & 183 are just two different provisional responses; 180 is ringing and 183 is Session in...

You can call json.decode in lua.

I believe you can, have a look at: https://freeswitch.org/confluence/display/FREESWITCH/mod_lua#mod_lua-CLIUsage:luaandluarun

OK, what I'm about to say may sound a little confusing, and I may not be strictly correct, but here we go... I can see from your packet capture that 206 (192.168.2.65) sends an INVITE to 207. 192.168.2.25 is the IP address of your Fusion PBX. Your FusionPBX receives the INVITE and responds...

Can't see the Wireshark attachment, did you upload it?

sngrep is a good tool, but it will not give you any more information than you are already getting from Wireshark. I would not go down the route of changing protocols at this stage. SIP uses UDP for good reasons.

You cannot set a channel variable this way because they only exist when the channel is created. But you could probably use a global variable: fs_cli -x "global_setvar testvar=hello" fs_cli -x "global_getvar testvar"

Then simply add it in to the FreeSWITCH dialplan XML.

The full Wireshark trace would be more helpful, so we can see what messages, if any, are being sent to the destination. Have you confirmed that both softphones are registered OK?

The simplest way would be to put it in your dialplan near the top along with call-direction, variables etc. Then the variable will get set for every call that passes through the dial plan. (Menu->Dialplan->Dialplan manager).

My apologies, I thought MicroSIP was a soft phone!

OK, from the menu choose Advanced->Access Controls. Click on the control named domains. Now add an entry... In the "type" drop down box choose "allow". In the CIDR box put the IP address of your SIP Trunk Provider followed by /32. For example 88.99.100.101/32 Do not put anything in the "Domain"...