mc_

I just see calls to extensions 6601 and 6602? What am I missing here? Limits only apply to call legs that involve a "resource" which is an upstream carrier for most folks' use cases. You'll need to dial DIDs to exercise the account limits

@huwsc90 your best bet is to look at FreeSWITCH's supported endpoints as all the audio muxing will be done there. Verto is an option for sure. Or you can do pure WebRTC+SIP using Kamailio as the wss server.

@louyingge there is not a way to track channels going to a particular carrier and use a different carrier if some threshold is reached. You can limit an account's channel count going to/from carriers. Could be an interesting feature if you wanted to sponsor it though!

@huwsc90 Its mostly in the form of JSON schemas for the payload formats. Most of the info will be in the source (which hopefully is readable for what you're wanting, even not knowing Erlang). For instance, a route request to setup a call involves the 'req' payload published to all bound queues, each of which will respond with a 'resp' payload if able. One 'resp' payload will "win" and receive a 'win' payload back on a direct queue. https://github.com/2600hz/kazoo/blob/master/core/kazoo_amqp/src/api/kapi_route.erl In there you'll see the 'req_definition' which will list the JSON keys (named headers) of the payload, the routing (and binding) keys used for publishing/binding, etc. https://github.com/2600hz/kazoo/blob/master/applications/crossbar/priv/couchdb/schemas/kapi.route.req.json In this directory, you'll see auto-generated schemas. "kapi.route" maps to "kapi_route.erl" for instance, and the 'req' payload. Should at least get you started, get a queue bound to AMQP for messages, and you can start inspecting things.

@Gagan KAZOO has no control over the format of the INVITE's RURI; that is generated by the sending side. Can you clarify why you need E164 format? What are you trying to do with the call?

If you built this from source, make sure all the deps/ built properly (should be listed in make/deps.mk). You can stop the release, do `make sparkly-clean compile build-dev-release` to clean everything out, fetch all deps, recompile everything, and build a development release. Then `make release` will start the dev release. Other things to check are that you're compiling with the correct version of Erlang (see make/erlang_version).

If this is a new system, yes, KAZOO should initialize relevant databases and configure the default list of apps to start automatically. You'll want to bump log level (typically syslog) in /etc/kazoo/core/config.ini to debug, then restart the VM and see what happens, why the default apps weren't started. Could be a temporary issue too (though I routinely reset my database cluster for dev and don't have issues with the default apps starting).

https://docs.2600hz.com/dev/applications/crossbar/doc/filters/

Hard to say why the other apps aren't starting offhand. Consult the logs from startup. You can manually start with 'sup kapps_controller start_app {APP_NAME}'

Hi @Beka Gelashvili When I'm testing alternative FreeSWITCH instances, I use: sudo FS_BIN=/usr/local/freeswitch/bin/freeswitch /usr/sbin/kazoo-freeswitch start -nonat This starts FreeSWITCH in the foreground so if you exit the console, FreeSWITCH will stop. You can use the -nc flag to put FreeSWITCH in the background and use fs_cli to connect that works better for you.

Are you looking for improved docs? https://docs.2600hz.com/dev/applications/pivot/doc/kazoo/ https://docs.2600hz.com/sysadmin/doc/install/configure_kazoo/ Are you looking for live training where something is built and you have an instructor to bounce questions off of? What's lacking in the above docs that is preventing you getting started? Or what was unclear? Suggestions for improvement? While I understand live instruction can be a boon, it is also not the most scalable (only those attending really benefit). Would a video tutorial of the basics suffice? Any tutorials you've done in the past (videos or articles) that really nailed getting you started? Also, how to identify where you (generic you) are starting from is really hard. For instance, a recent post asking how to do things outside of SmartPBX led me to believe the person was familiar with cURL or postman which turned out to be a false assumption. I've led KAZOO installation and management trainings (designed for sysadmins familiar with Linux) where folks joined who only knew Windows (and not sysadmin-ing of Windows)...what should the trainer do in that situation? Identifying the prerequisite skills for a training, and communicating those to potential trainees isn't always easy. We assume a lot of our knowledge is known by others because "its obvious" or "how could someone *not* know this" All this to say, I would love to improve the ability for folks to get comfortable using the APIs, Pivot, websockets, etc. But I believe I'm too far gone from beginner status on those things that I don't know what you all don't know, if that makes sense So I'd love feedback on what's missing from the docs as well as concrete suggestions for what to add (those docs are all backed by git repos on Github - which again is an implicit assumption that someone with suggestions will 1. have a github account 2. be able to edit the doc in question 3. be able to submit a PR to change things).

Hi Alex, First, I would recommend https://docs.2600hz.com/sysadmin/doc/install/install_via_centos7/ I'm not sure if the powerpbx guide is still updated or not, but our guide covers similar ground and is updated as folks encounter issues. It is backed by a public Github repo as well for all to contribute. As for your issue, KAZOO (and most SIP setups) use the domain portion to figure out what account to associate with the caller. In your case, it is unlikely you created an account with a SIP realm of Kamailio's IP address. You will need to configure zoiper to use the SIP realm of the account and, if the domain doesn't resolve to Kamailio's IP address, you will need to configure zoiper to use an outbound proxy (or zoiper's equivalent) to point to Kamailio's IP. You can read https://docs.2600hz.com/user_guides/doc/manual_phone_config/zoiper/ Might help; adjust realm and outbound proxy to your setup

HI @vageesh welcome! It is hard to say but it appears that you've misconfigured the API url. Your API responds with an appropriate error with 'curl -v -X PUT http://143.110.176.191:8000/v2/user_auth' So check your monster config.js and ensure the URL is similar to this ^

The process is straightforward with ffmpeg. KAZOO does have some normalization routines, mostly for mp3<->wav normalizations. You could certainly sponsor work to support more source formats to normalize.

audio/x-m4a is NOT a valid mime type: https://www.iana.org/assignments/media-types/media-types.xhtml As far as I can tell, it is proprietary to iOS. Hard to say if FreeSWITCH supports it - likely need to use mod_vlc to do so which I don't believe is installed typically.

You would set the callflow's "numbers":["11"] with a flow of {"module":"resources", "data":{"to_did":"{NUMBER}"}} (as an example).

You may need to add devices, create callflows for those devices, etc. See https://docs.2600hz.com/sysadmin/doc/install/configure_kazoo/ Or load monster-ui and smartpbx to set that stuff up if you prefer the UI route.

Revisit the kamailio section: https://docs.2600hz.com/sysadmin/doc/install/install_via_centos7/#setting-up-kamailio And check that you've configured things as explained in the doc.

Always nice to get a win!

Hi Ash, First, welcome! Glad you're here! I would start by looking at the SDP in the A-leg's INVITE to see codecs offered, as well as what codecs are then offered to the B-leg from FreeSWITCH. Could be a transcoding issue or misconfigured codec selection. Payload type 0 is typically PCMU but you're log shows PCMA being used. Perhaps look into what codecs FreeSWITCH thinks are loaded? `fs_cli -x 'show codecs'` should list `codec,G.711 ulaw,CORE_PCM_MODULE` among others.

You can disable call_waiting on the device or user config: https://docs.2600hz.com/dev/applications/crossbar/doc/users/#call_waiting

@Shah are you creating one connector doc per PBX or one doc for all the PBXes?

@Pargat sounds like, unfortunately, a lot of spam and malicious traffic originates from various countries; operations has blocked large swaths of IP blocks to combat it and you appear to have been caught up in that. We're working on getting a mirrored host up so that folks like you can access the repos without needing to resort to a proxy or VPN. More to come!

Not aware of any restrictions (asking around). Can you try fetching that URL with wget or cURL and see what they say?

You cannot, SUP is only for the cluster administrator. What are you trying to do as a sub-account that you need them to use SUP for?