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martin

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Posts posted by martin

  1. 20 minutes ago, fmateo05 said:

    I was previously inviting some asterisk/issabel users to switch or migrate to 2600hz's kazoo platform. Also telling them about a basic installation and configuration.

    They  find it 'difficult' and return back to their simpler PBX.  

    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?
     

  2. On 11/6/2019 at 4:10 AM, Guillermo Prado said:

    Hi, this is an issue with kazoo community, because the support is very poor and the documentation is outdated, I think is very hard to do something with kazoo 

    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

  3. On 8/17/2019 at 7:15 PM, Darren Schreiber said:

    There are multiple scenarios here:

    1) Shoutcast stream doesn't stream at all. Right now when we start playback of an audio file we wait for it to start which is confirmed by various events. These events never fire, so code doesn't advance.

    2) FreeSWITCH itself is actually where the main issue is - it's default behavior is to either advance or hangup, depending on the situation, when you ask it to do something invalid like play a non-working stream. In addition, it's behavior halfway through a stream when a connection closes is to advance, which triggers events that cause us to think the call should advance, further gumming up the works.

    3) Neither FreeSWITCH nor Kazoo have a concept of an alternate stream to play (even silence is a stream) so we'd have to somehow update the code to support that

    4) You would likely have to make the failback strategy configurable, thus a GUI component. Not everyone will accept just "silence" as a failover strategy and will complain just as loudly here when their callers suddenly hear silence that it's broken/not usable.

    All the above would of course have to be tested.

    I suspect it'd be about 40-120 hours of work (one to three weeks of hacking on this basically), with knowledge required being a mix of C code and Erlang. Then about 20 hours to test each scenario. You also need to get the C code changes committed upstream to FS since we keep in sync with their branch now, so there's about 10-20 hours of "management" in there. So the entire thing is probably 80-200 hours. Assume $150/hour minimum for this skillset, it's a $12,000 - $24,000 request.

    One way to fund this is we could get upfront commitments for people who want to use this. Let's say 10 resellers committed to pay for it for a year at $100/person/month, or up to two years to cover the higher end of the estimate. That would cover the cost.

    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



     

    On 8/17/2019 at 10:21 PM, Darren Schreiber said:

    I'd be down setting up some sort of system to fund these requests. Let me see what I can come up with.

    oh missed that)

  4. On 12/12/2018 at 1:34 AM, mc_ said:

    I think you need to use `ledgers` for "funny money" and transactions for "real money" stuff.

    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

  5. On 2/13/2019 at 6:30 PM, eurovoip said:

    I have 100 DID's not assigned but available in one zone (IP ) and 10 assigned to an account in Kazzo. and one day the zone goes down and the user can connect to zone B only where  those IP's are not even assigned (because they where in assigned to the zone on's IP ). Unfortunately my provider doesn't let me point the DID's individually. 

    How can I make sure the DID's will work on any of the two clusters ?! 

    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:

    “The SRV RR allows administrators to use several servers for a single domain, to move services from host to host with little fuss, and to designate some hosts as primary servers for a service and others as backups.”

    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. https://2600hz.atlassian.net/wiki/spaces/docs/pages/75464706/Install+Nagios+monitoring

     

    Nagios XI

    Service Status Dashboard Nagios XI 2014

    Nagios® XI is the most powerful IT infrastructure monitoring solution on the market. Nagios XI extends on proven, enterprise-class Open Source components to deliver the best monitoring solution for today's demanding organizational requirements.

    Designed for scalability and flexibility, XI is designed to make problematic IT monitoring tasks simple, while retaining the powerful attributes of its enterprise-class foundation blocks.

     

    Important

     

    Important: Nagios Enterprises highly recommends and will only support installing Nagios XI on a newly installed, “clean” system (a

    bare minimal install with nothing else installed or configured). Centos 6x does the job

     

    Step-by-step guide

    Please follow these simple steps

    1. cd /tmp
    2. wget http://assets.nagios.com/downloads/nagiosxi/xi-latest.tar.gz
    3. tar xzf xi-latest.tar.gz
    4. cd /tmp/nagiosxi
    5. ./fullinstall

    Enjoy your Nagiosxi install

     

    You may want to add a firewall and start configuring your setup to monitor  your cluster, a free license covers 7 servers!

     

    After installing you can add different plugins specific for SIP, Kamailio, and so on.

     

    You need to install a Agent that will connect your servers and Nagios

    This document describes how to install the Linux monitoring agent on target RHEL and CentOS Linux servers. Other Linux
    distributions may be added in the future.


    Target Audience
    This document is intended for use by Nagios Administrators who wish to monitor Linux servers with Nagios XI.
    Installing the agent
    Download the Linux NRPE agent to the /tmp directory on the Linux server you wish to monitor.
     cd /tmp
    wget http://assets.nagios.com/downloads/nagiosxi/agents/linux-nrpe-agent.tar.gz
    tar xzf linux-nrpe-agent.tar.gz
     cd linux-nrpe-agent
    ./fullinstall

     

    http://support.nagios.com/

  7. 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

     

  8. 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 ))

  9. 5 hours ago, omer said:

    Hi all,

    I am new. I installed kazoo on my VM and am trying to test it with soft phones. However, in monster ui, I cannot add neither any number or device nor any credit card. Credit card button does not exist, and add number from spare part cannot be pressed. Only option is Buy Numbers. It comes with one country option which is US and with no number list. I need some options to connect it with soft phones and then apply some call flows. But no chance. Is there any options to do this without buy the commercial version.

    Regards.

    Hi,

    I feel your pain ))

    To get the creditcard tab in accounts app read this, in short: A payment processor needs to be active and reading that  thread will help u.

    It will ennable the creditcard tab as its quite logic its not there if no payment processor is active.
    Adding a number is a different story, but it does not work because u did not activate or configure a DID buying provider (or carrier)
    How to do that? Well Kazoo by default supports bandwith and voip innovations and perhaps also telnyx although i heard its not working (i did not test)

    To activate VoiP innovations, please read this.

    The number default country... well i asked the same, but no answer yet. It could very well be though that after u activate the DID number provider/carrier it will magically work.

    That u can not add a device seems very very strange, and i actually do not believe u ))
    Can u tell me what happens when u try to add a device, and upload some screenshots of the process?

     

    If u found my answer helpfull, please click on the hart icon, it makes me more popular ))

     

  10. i tried and got this response

        ... ........
        ......+OO=..
        ....IOOOO8?......
        .:O,.8OO8OO7......
        ..IO.~OOOOO8O.. ..
        ...O8.OOOOOOO8~... .. .
        ...I8=,8OOOO8OO=........
        ....$O.:8OOOOOOO?.......
        ... .8O.OOO8OOOOO8......
        .....7O8.ZOOOOOO8O8,....
        .. ...88I.8OOOOOOOO8:....
          ....:OO7,OOOOOOOOOO:...
              .=OO,$8OOOOOOOO8Z...
              ..$O8.=OOOOOOOOO8O,.......
              ...O8O.ZOOOOOOOO8OO,......
               ...OO8.$OO8OOOOOO8O:.....
               ...?OOO.8OOOOOOOOO8O$...     ..
               ....8OOO.OOOOOOOOOOOO$.... .  . .
               .....8OO7,8OOOOOOOOOO88.........
               .  ...OO8:,OOOOOOOOOOO88..... .
               . ... +8OO++OOOOOOOOOOO88,.....
                ......~8O8.:OOOOOOOOOOOOO.. ..
                    ...OO88,$OOOOOOOOOOOOO?...
                      ..8OO8.OOOOOOOOOOOO8O+.... .....
                      ..:88O8.OOOOOOOOOOOOOO+...............
                      ...,OOOO.OOOOOOOOOOOOO88.:O8OO8O:.....
                       ...+OO88.O8OOOOOOOOOO8:Z8OOIIO8O8....
                       .....OO8O.8OOOOOOOOOZ+:8OO8~...=OOZ..
                        ....:O88O.8OOOOOOOO?.?8OOO88~...8O:.
                       ......$OOOO.OOOOOOO.O.$O8O8O8O:..~8D.
                          . ..?OOO?.8OOOO~.8,+OOI:OOO8...7O.. .
                             ..ZOOOZ,8OOO..8+,OO...8OO...:O: ..
                             ...7OO8~.8O8,.88.8O:..8O8:..,8+...
                             .. .ZOOO+,OOO.$8~OOOOOOOO~..+8I..
                             .....8OOO,=88$:88,8O8OO88...8O,.
                              .....OOO8.,OO=8OZ.O8OO8...8O8.. .
                              ......OOO8.I8OZ7O8.8OO?~O8O8,...
                              .  . ..$OO8.~OOOOOO.+OO88O?...
                                   ...88OZ.=OOOOO88:....,.....
                                    ...8OO8.OOOOOOOOOOOOOO...
                                    ....ZOO7.?8OOO8OOOOOOO+..
                                     ....?OO$.=8OOOOOOOOOOO~...
                                   .......,OO8.,OO8OOOOOOOO8:..
                                     ..... ~OO+.+88OOOOOOOOOO.......
                                    ........=8O$.IOOOOOOOOOOOO... ..
                                          ....OO=.~8OOOOOOOOOO8. .
                                          .....OO,.78OOOOOOOO8O7....
                                           ... .O8+.~OOOOOOOOOO7....
                                          .......D8..+OOOOOOOOO.....
                                           . ... .?8~.OOOOO8O8......
                                           ...... .?8:.?8OO8+.......
                                             . . ....8..I$........
                                            . ..   ............  .
                                                     ....

    With some fantasie it might look like a kazoo, but is not a auth token

    The base url, is that only server name, or also with http:// and with the /V1 ??

     

    image.thumb.png.318f4fb2dc5946c426cc8370079b3b87.png

  11. 20 hours ago, amn said:

    If using IP authentication, the carrier is associating the IP with the customer account on their end.  So on the carrier end, you cannot have more than one customer account with that carrier using the same IP.  Otherwise the carrier would not know who to bill.  That is for termination (outbound calls to the carrier).  For origination (DID inbound) I am not sure if it would be a problem or not since the carrier knows which customers are using which DID's and should not care what IPs they are assigned to.

     

    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

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