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Showing content with the highest reputation since 07/06/2017 in Posts

  1. 3 points
    So if you have a free Vimeo account, you can add the video to a post, by copying the "normal" link (not embeded) into the post, and the forum software will allow for playback from the forum. Only thing, is a touch of suggested video's at the end of the video playback. @Rick Guyton
  2. 2 points
    Cool! Is it possible to add the intercom key type option to SmartPBX as well?
  3. 2 points
    Hi"! Well as i am a captain by profession and do this to learn it sometimes goes over my head )) meaning that if a problem becomes more complex, and lets say has mmore then three or four levels, i get lost. I cant see the big picture anymore. So thats why im a good fir for writing simple docs, but in a workflow kind a way. Per example: Here is how to acieve this step: 1. do this 2. and that because of this or that 3. etc U can see the style in here, here, here and here. I remember that one of the sad things was, that at that time the idea of dealing with docs in a different way was new and just being pushed a bit. But for some things i needed help, and it was not available. of course, being naive and enthausiastic u want to move more fast then what is posible sometimes. And expected a bit too much from 2600hz perhaps, but... now i see this new forum, a lot of new faces and feel some more energy here )).
  4. 1 point
    Hello, I like the new interface/app you created for this. It will make it easier to train and does not require advanced call flow access. Some feedback: It would be nice if we could specify or enable a feature code for people to automatically add the last received call to a default blacklist.
  5. 1 point
    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
  6. 1 point
    Excellent job organizing and executing this event. It was worth the time to attend! We learned a lot and made some excellent contacts within the Kazoo community. Will the presentation material be made available to the attendees? There was a lot of good information presented and some of it came faster than my note taking skills could capture.
  7. 1 point
    Super duper admin is the primary account you create and all it's users, by default. It gets created with your first account automatically. For bulk import of numbers, you can bulk import numbers with the CSV Jobs app I believe. For "No, I don't think so? What I mean is a simple sip account, i.e. a telephone number user/pass with a credit limit. so that an end user i.e. Joe Homeuser can purchase a phone and a $10 monthly service, but has no PBX attached from his perspective. " I think what you're saying is you want to create accounts with just one phone/user in them and then have limits in them, like a calling card service or home phone service. This is possible but there's no "one button push" to do it. Everything is API based though so you could make your own app to do it.
  8. 1 point
    Hi there! Thanks for trying out Kazoo! Lots of questions indeed. Let me try to answer each, below... We are looking for assistance on the documentation side, if you are interested in helping out, please let us know! So actually the rating stuff is kind of generic and you can use any currency you like, but we hardcoded US in the UI for now. So you would just have to change that. We don't deal with tax right now, and the billing engine is kind of archaic and is due a re-write. I would recommend using only the limits and per-minute rate tracking features and overriding the UI to put in your own currency. We would welcome contributions to make the currency dynamic, we just haven't had the time to do it. Like a single install? Just remove accounts manager from the apps list and it's a standalone PBX if that's what you mean. We have some more open-source apps coming out soon but not yet. The provisioner / carrier manager / dialplan apps are currently closed source. Please contact sales@2600hz.com for those. There is a blacklist application that I believe is released open-source. You can use that. Thanks! Go into the Futon database and go into the accounts DB. Delete the bogus account. Also, drop the databases for the bogus account as well. Then you should be good to go.
  9. 1 point
    Thats true, one could sent a 1. specific header (or how we call that, flag?), depending if the carrier allows it of course. 2. Add an ip to your system and ask a Kazoo guru to explain how to force that ip to be used for this specific account. Inbound is no issue, but if Kazoo allows IP auth, and customers can bring their own carrier, then the risk is rather high to have duplicate ip adresses. If not 100% guarunteed. If u have 1000 clients, this problem will exist. U could do smth with outbound CLI but that also depends on the provider and could be spoofed. I dont know how Kazoo handles this, but my brain does not see a solution other then stated
  10. 1 point
    Thank you Bobby for the referral for Judy at Telecom Professionals, Inc.
  11. 1 point
    Hi, we're a small startup hosting PBX for our clients and we are looking into Kazoo for our VOIP needs. So, what we need to know is: 1- Will a self hosted Kazoo install have the same benefits of your SaaS? 2- Does Kazoo do LCR, Rating, Billing, Accounting in one or more of it's apps? If not, how hard could it be to implement these functions on Monster UI or externally? 3- Is it possible to acquire the closed/non-public apps on our self hosted installation? 4- What is needed to localize Monster UI and the dial prompts to our language? In any case, we could PR back our changes to 2600hz. Thanks.
  12. 1 point
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