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Looks like you're missing a single quote in your lua curl command, however it is not missing in your cli curl command, as illustrated in following screenshot; Try adding a single quote before the "application/xml" part, as follows; cmd = " curl " .. api_url .. " -H \"Content-Type...

I am guessing the issue is likely related to escaping the string when used in the lua script. Because in the lua script the cmd is within double-quotes, example; cmd = "curl -u ....................... <response>0</response> </sms> "; This means if you need to use any double-quotes within your...

Hard to see what is going on from the screenshot, but the error suggests the receiving server is unable to parse your data as valid XML. Start by confirming your curl command is correct, and you have properly quoted/escaped everything. If you post your code from the...

To test the curl command manually, use something like this and replace the variables (signified by a $) with your test values. curl -u 'api_url' '<sms> <user> <username>$username</username><password>$secret_key</password></user><source>$outbound_caller_id_number</source> <destinations>...

As far as i know, there's no real sms log other than which is shown in fs_cli. I would start out by confirming your curl command works successfully, does it work if you manually run that curl message from cli? My guess would be your strings are not escaped/quoted properly. Try something like...

To add a new carrier you need to add support for sending and receiving via that carrier. Sending via new carrier: You can edit `/usr/share/freeswitch/scripts/app/sms/index.lua` to add a new carrier, around line 400~ you will find a bunch of if-else statements. Add a new if-else for your new...

The instructions at https://github.com/fusionpbx/fusionpbx-apps/tree/master/sms are still relevant and worked fine for me recently. Where are you getting stuck?

Adrian is correct, you can't multicast over the internet. You could however tunnel multicast over the internet via a vpn, or as Adrian suggested use some kind of relay to forward/replicate the packets.

Glad you figured it out. As you mentioned, "user_group_uuid" was the key we were looking for. You can either hardcode your uuid in the ldap.php file, or set it up to use a config variable as the uuid as per one of my last posts. Be sure to undo/remove any test/debug stuff you added along the...

You can get the group uuid by going to Advanced > Group Manager > click the group you want, copy the uuid from the url It will look something like this;

When looking at that ldap.php file, it appears the user is assigned to a new group as per this line; $array['user_groups'][0]['user_group_uuid'] = uuid(); You could try replacing that line with; $array['user_groups'][0]['user_group_uuid'] = (!is_null($_SESSION["ldap"]["assign_group_uuid"]) &&...

I have multi-level IVR for at least 10 domains, no issues as of yet. Yes it is possible to create loops, it is up to you to handle loop scenarios and provide callers a way out.

Based on the following link, it appears the ldap user is assigned to the `user` group like you want, assuming that username doesn't already exist in fusionpbx...

I see. That explains why I couldn't find any such docs. Thank you

Hi all, I'm wondering where I can find a list of available icon names for Yealink phones? I'm specifically looking for models T46s and T56a, but any pointers would be appreciated. I'm trying to change the icons shown on the phone for the following setup;

You're entirely correct. Can't believe I missed that, I figured it was somewhere obvious. Thanks again, I appreciate you

Hi all, I have a call center queue using ring-all with no tier rules, which works fine and calls all available agents. The issue is that it rings each agent for only about 10 seconds, can anyone advise where I can change this value? The agent's extensions have a "Call Timeout" of 30.

You could look into uuid_broadcast which seems to be async, you can use it to execute another dialplan. The examples in the docs look promising; https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-uuid_broadcast

Just curious what system they were working with prior to switching to Fusionbpx? In my limited experience, Cisco voip phones are a pain in the ass when it comes to SIP.

It is possible its either your firewall or a firewall the customer is using. I guess it couldn't hurt too much to whitelist all the traffic from their IP to see if that helps, if so then go from there to further refine the access. CSF is probably a bit overkill for a simple voip server, but if...