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Hey @Agha Assad , have you been lucky with this? I am trying to do the same but I cannot figure out how to do it for an outbound call. For incoming calls it is pretty straightforward. I thought it was something like bridging the call and then playing back the audio but it does not seem to be...

@toolfolks , log in to the Linux server console. Type fs_cli. What do you see? Try restarting the GW, what do you see? Did you add the ACL rules to allow connections from your provider? Follow this.

Ahh... I am not sending ogg over the wire! That is very true. I was mistaken in thinking that given the ogg file was smaller the audio used by the RTP was also smaller but that is of course not the case! I only have a bunch of audio files, disk space is not really a problem here. Thank you all...

Hi all, Question: is there any performance difference in using WAV, MP3, OGG files (at 8K) in an IVR (for playing back audio to callers). Clearly ogg files are more efficient from the band width perspective but would I be paying any penalty on CPU or memory processing? I cannot really find any...

As far as FusionPBX is concern the GW is openSIPS. So, you just need to send the calls there. When they get to OpenSIPS you need to tell it to forward the calls to Teams. 1. Create a SIP trunk (a gateway) between FusionPBX and openSIPS. 2. FusionPBX makes it really easy to create the outbound...

Thanks @ad5ou . My need to restrict ports comes from a third party recording system my customer has. Since FusionPBX/Freswitch do not have a CTI interface to speak of, the only way to control which is which is by RTP port range. It is a long story but it would help if I could know that ext1xxx...

Thanks @ad5ou for the reply. Just to double check, the configuration file to modify would be /etc/freeswitch/autoload_configs/switch.conf.xml, right? I am just asking because when I search for it in my Fusion box I find seven of them, however it seems to me all others are either buck-ups or...

Hi all, I was wondering. Is there a way to set a specific RTP port range per domain? I need to create some firewall rules in my edge router and it would really simplify my life if I knew the RTP port range per domain. Thanks!

Fantastic. Thanks very much @gmaruzz !

Erlang seems too complicated to me. I think Elixir would be a good fit: the power of Erlang plus the flexible code experience of Ruby.

Awesome review! Thanks for taking the time @brb5548. As I mentioned before I am not a developer so I am not sure about all the architectural differences in the languages but coming from a Telecom background I have a lot of confidence on Erlang, it is multi thread and HA is its DNA. It's been...

CTI is a concept. Each telephony vendor has its own approach. For instance Genesys T-SAPI or SIP server, Avaya TSAPI. Cisco and Microsoft have been using TAPI which is, as far as know, the closes to a standard protocol you can get. I have been working, as an integrator, with CTI powered system...

Excellent link @babak . I wish I had found it a few months back. We had a bid for a contract at a government institution which needed exactly that, a way to connect their existing Avaya switch to Teams without expending a fortune in the process. We ended up comparing AnyNode and Audiocodes...

So, one can create its own CTI through ESL, right?

@Henry Jones , it does not seem FusionPBX to be the problem. The gateway provider is reporting a unavailability error. Can you check the logs on 222.222.222.222? Those will likely narrow down the issue. I do not see any problem on the fusionPBX side.

ahhh I get it! Thanks @Adrian Fretwell.

@kiran870 , you are getting a permission denied error to read the require.php file. By default that file should belong to the www-data user which should have read permissions. In fact the default permissions give read access to everyone. Have you modified those? Have a look to that file's...

Hey @Adrian Fretwell , for my own education, what is the advantage of using bridges instead of using regular dialplan destination rules? Let's say one fusionPBX has extension range 3xxx and the other 4xxx. Cannot I simply create a SIP trunk between the two and then tell the dialplan to route...

Hey @DigitalDaz , I was thinking, a CTI provides events and integration into the system. Isn't it what ESL can provide? I know, it would be all custom code but I was thinking about getting started on such a project. What do you think? I am more of a telecom guy than a developer, so that...

Run a tcpdump on the local server. Check it out with Wireshark or sngrep. Here is a useful guide for tcpdump