martin

If u want to have the developer app back into Monster, use this: https://github.com/yumminova/kms-ui/tree/master/apps/apiexplorer Needs updating to latest monster version though if thats what u are running

I see, but if u would give them a bare asterisk box with external Gui they would have the same. Try bluebox, it's free and Kazoo evolved from that project

Billing Services (and service plans) have received a major overhaul in 4.3. APIs have changed for interacting with services so if any one has built on top of the `service_plans` API, they will need to take note and adjust their code. Cant find updated docs on this

Totally understandable, bugt it says very little about Kazoo. I have ther same problem when i switch from ms Paint to Photoshop. Now, im having u grumpy day, so i disagree with everyone )) butof course there are valid points made here, but in the ned there are many options available to everybody. Buy support, go hosted, or learn. The lacking docs.... thats my pain as well, but 2600hz owes me nothing, so i try to be nice without kissing ass, write some howto's, help some forum members and hopefully get an answer . If we all did that it would be such a nice forum to be that the dev team would stop hiding on IRC and hang out here... Just curious... why did u invite asterisk user to switch?

As long as you wont get a support agent or dev teal on it, its useless, unless you want to say hi, but phonenumbers and sip uris are available on this forum most of the time.

Support is not poor, just not free if u need more then this forum. Now... is it very active? Nope, but thats because if u use Kazoo on your own, you are probably very well informed on the used techniques. I do agree u need a lot of specific knowledge, and read your @ss of, but thats just how it is. Working with the APIs is not hard, it sometimes is a bit messed up because of that old docs, but normally u will be able to find your way. Your options: Buy a support package, go Hosted , read up and learn

i think we spoke about this a few years ago, maybe create a infrastructure for gathering finances for these kind of developments, and perhaps other requests? Its probably utopia to get many people to chip in, but then at least it clear. Gofundme style I would pay for what i want, but dont have a big budget, its not even a business yet. But chippping in for interesting development would be smth i would do oh missed that)

can u explain the use case for funny money and real money? I was using kazoo in 2016/17 and am trying to get back up to speed

u normally use srv records in your dns to map a single domainname to multiple ip endpoints (your kazoo installs). If u have different zones and one is unreachable your calls and clients will be automaticly mapped to an other zone. If your did provider zallows you to put a domain name as destination u should be fine, if its only an ip address, your provider probably has a fallbasck solution, such as different destination ip addresses as destination. I hope i understood ur question. About srv records: borrowed from https://www.onsip.com/voip-resources/voip-fundamentals/dns-srv-records-sip What are SIP DNS SRV records? Let’s say your name is Al and you work for a company called Acme Corp. Your company has recently signed up for a corporate VoIP phone system and you’re given a SIP address, Al@acme.com. In an ideal world, your friends and colleagues who are also familiar with VoIP know that they can dial Al@acme.com to reach you wherever you have a device registered. How does this work? Dialing by domain names allows you to have a public SIP address that follows you much like your email address would. It doesn’t really matter where you are or what device you use; as long as you’re ‘logged in’ (registered), SIP communications will be redirected to your current location. A Domain Name Server (DNS) SRV record for SIP does the same thing as a mail exchange record for email. When someone calls you at Al@acme.com, the SRV record tells their SIP phone to do so by connecting to the domain belonging to your VoIP phone system (SIP) provider. This does two things for SIP providers and SIP users: 1. Greater stability From the RFC: If for some reason the ‘host’ with the highest priority cannot be reached, the SIP phone or proxy trying to reach the user within the domain will attempt to reach the next host defined within the SRV record. 2. Allows SIP users to get their own domains for their SIP addresses, regardless of the domain of their SIP provider We refer to this as SIP service or SIP hosting, and it’s a feature of an OnSIP account. OnSIP runs several SIP proxy servers, which can handle SIP users in multiple domains just like a mail server handles e-mail for multiple domains. SIP DNS SRV record settings To use OnSIP's SIP hosting, the zone file in the DNS SRV records of the user’s domain (acme.com) needs to point to sip.onsip.com. Calls to the user's domain will then be automatically redirected to an OnSIP SIP proxy. Here is what the DNS SRV record looks like: Service: SIP Protocol: UDP Name: acme.com (your domain name goes here) Priority: Weight: Port: 5060 Target: sip.onsip.com TTL: 1 hour Example DNS SRV record The following is an SRV DNS Lookup of sip.voice.google.com. As you can see below, the query returns 5 hostnames for inbound SIP traffic to that domain. Google Voice UDP Record $ dig _sip._udp.sip.voice.google.com SRV ; <<>> DiG 9.6.0-APPLE-P2 <<>> _sip._udp.sip.voice.google.com SRV ;; global options: +cmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 8463 ;; flags: qr rd ra; QUERY: 1, ANSWER: 5, AUTHORITY: 4, ADDITIONAL: 9 ;; QUESTION SECTION: ;_sip._udp.sip.voice.google.com. IN SRV ;; ANSWER SECTION: _sip._udp.sip.voice.google.com. 86400 IN SRV 20 1 5060 alt1.voice-sip.l.google.com. _sip._udp.sip.voice.google.com. 86400 IN SRV 10 1 5060 voice-sip.l.google.com. _sip._udp.sip.voice.google.com. 86400 IN SRV 50 1 5060 alt4.voice-sip.l.google.com. _sip._udp.sip.voice.google.com. 86400 IN SRV 30 1 5060 alt2.voice-sip.l.google.com. _sip._udp.sip.voice.google.com. 86400 IN SRV 40 1 5060 alt3.voice-sip.l.google.com. ;; AUTHORITY SECTION: google.com. 146471 IN NS ns3.google.com. google.com. 146471 IN NS ns2.google.com. google.com. 146471 IN NS ns1.google.com. google.com. 146471 IN NS ns4.google.com. ;; ADDITIONAL SECTION: alt1.voice-sip.l.google.com. 300 IN A 74.125.95.192 voice-sip.l.google.com. 300 IN A 74.125.95.192 alt4.voice-sip.l.google.com. 300 IN A 74.125.95.192 alt2.voice-sip.l.google.com. 300 IN A 74.125.95.192 alt3.voice-sip.l.google.com. 300 IN A 74.125.95.192 ns1.google.com. 342957 IN A 216.239.32.10 ns2.google.com. 342957 IN A 216.239.34.10 ns3.google.com. 319271 IN A 216.239.36.10 ns4.google.com. 342957 IN A 216.239.38.10 ;; Query time: 18 msec ;; SERVER: 207.172.3.8#53(207.172.3.8) ;; WHEN: Fri Mar 11 18:01:49 2011 ;; MSG SIZE rcvd: 494

Not meaning that i believe its not comming! just.. dont wait for it

I would not wait for it, some are waiting for a few year already

Hi Im writing this while feeling in doubt if its not already posible, if so please forgive me, ill delete the thread. I would like it very much, if the available credit on a kazoo install could be used for all costs, so for calling, but also for devices etc. As we can define allow prepaid in the limits doc, i somehow expect it to be available but cant find how to set it up. It would really simplify things for billing for my specific usercase

thanks! im geting old )) working like a charm! Thanks for these instructions, a lot!

Changing )) haha that's old news, but it would be cool. I did see the http bookkeeper, and actually that's almost enough.

So, i assume that u are new to Kazoo and have installed a single server. Please correct me if i am wrong. First steps for testing: make sure u have a domainname for your system and that the dns is setup correctly While logged in as admin, create a new account by visiting the account app. (its better to not "work" in admin account) Then masquerade or log in as that account and create a device in the callflow app, or in VOIP app, and with that account data register a phone. It should just work, if not look in the log files for Kazoo and kamailio, log files are situated in /var/log/kamalio and /kazoo If u want to call to the outside, u need to setup an outbound trunk (can also be called provider, carrier, sip trunk). I suggest that u set it up in the admin account and configure your sub account to use the system wide carriers https://2600hz.atlassian.net/wiki/spaces/docs/pages/79233046/use+the+Global+Carrier+offnet (my old contibution, deals with Kazoo UI, but still informative)) https://docs.2600hz.com/sysadmin/doc/config/global-resources/ https://docs.2600hz.com/sysadmin/doc/config/global-resources/ If u found my answer helpfull, please click on the hart icon, it makes me more popular ))