Recent content by bkcberry

One last question. Blind transfer works as expected, but attended transfer does not. How do I allow an external number? It looks like it is only looking up internal extensions.

Is that the only place? I've enabled that and rebooted just for good measure but still nothing happens when i type *1 on the softphone. I've attached the log from when it received the DTMF tones, so I at least know that's not the problem Edit: i got it working, had to move it further up in the...

Hey guys, i'm new to fusionpbx/freeswitch, trying to switch over to it from freepbx and asterisk so that i can take advantage of multi tenant. One thing i haven't been able to wrap my mind around yet is the dial plan, and specifically how to use it to transfer a call. On freepbx you can use the...

thanks again. for anyone else that stumbles upon this, i added privkey.pem to the end of fullchain.pem and saved the whole thing in /etc/freeswitch/tls as tls.pem and i'm now able to register and make calls using TLS and SRTP i also added lines to the end of my cron job to automatically do that...

awesome, thanks! one more question tho- can i use the letsencrypt certificates or do i actually need to generate my own? i assume i'd need to add some stuff to the renewal script to copy and rename the renewed certificate every 3 months. Just not sure if letsencrypt creates the type of cert...

hey guys, sorry to revive an old thread. Was just wondering if someone can point me to some instructions for setting up sip tls and srtp? I have a letsencrypt certificate set up and working for nginx, but from the bits and pieces of info i'm finding i can't get tls to work for sip.

Well i ended up rebuilding the server and for whatever reason it's working now. Thanks guys Edit- i think it may have been because external_rtp_ip and external_sip_ip weren't set to the server's public ip address

oh sure enough, i checked the sip logs on my telnyx's site and it says the 488 is coming from them. I opened a support ticket to see if they can give me any hints. I have a freepbx server set up thru these guys working perfectly, not sure if it's fusionpbx or something i'm just overlooking... i...

Is the 2nd leg the part of the connection between the server and the trunk provider?

Also, what did you mean by getting a log from the other endpoint? The sip trace I linked to in my first post is from gs wave.. I'm pretty sure the problem is between the extension client (gs wave) and fusionpbx because as I said earlier conferencing is working perfectly when calling in from the...

I tried TCP with only PCMU, same result. I'll get a more detailed log and see if it gives any clues. Thanks guys

Unfortunately i don't think it's either of those, in the sip client i have enabled all supported codecs (PCMU, PCMA, G722, OPUS, G726_32, iLBC, GSM, & G729) and i have tried changing the SRTP settings from disabled, to enabled/not enforced, and required. None has made a difference

here's the server log from a call just like the one in the sip trace above:

Hey guys, i'm new to freepbx and i'm having a problem getting an extension up and going. I have setup a conference and can call into it and have 2 way audio, so i now everything is working correctly with my gateway/trunk. The problem is when i set up an extension and connect to it with a sip...