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Posts posted by extremerotary

  1. Just now, Darren Schreiber said:

    Actually we are already finishing testing and planning to release the dynamic address feature. It's not just a bandwidth.com thing. You can pass the geo-location lat/long or a full address at the time of dialing 911.


    I figured it wasn't just them. You're doing a huge service for your customers and making it much more cost-effective for clients to go after hotels, schools, and some retirement homes with this type of feature. Awesome to hear!

  2. Just adding my 2-cents in here; It's not up to 2600hz to build this. Every carrier charges VoIP providers on a per-location basis. So even if 2600hz built something to make it look like it was all one, they would be getting charged on the backend so that wouldn't make sense from a business perspective. However, i know for a fact that since Ray Baum's Act, Bandwidth,com (3 day outage, i know i know) has been actively working on a dynamic E911 addressing service. In this way, 2600 could pass additional, more specific data to the PSAP using SIP headers (Like "room 121") instead of requiring an e911 registration per room. Last I heard they were close. Might be ready at this point. Bandwidth wasn't the only carrier looking into this, so since 2600hz aggregates multiple outbound carriers, i'm sure they are working with one to develop this dynamic e911 farther and bring that to their customers!

  3. On 9/23/2019 at 1:27 PM, safarov said:

    Error 488 Not Acceptable Here is related to used codec and encryption.
    Not IP used on caller side (private or public).
    Please paste INVITE here and codec related variables from "freeswitch.xml" 

    The error was that there were no valid candidates in the SDP because they were all local IPs. By applying the candidate ACLs, FreeSWITCH handles the SDP differently and is able to determine the public address to send the SDP. 

  4. Hey guys,

    I see that you've moved from using the sipjs stack to using the jssip stack for the libwebphone. I cloned the repo, set up the basic wss and tried to make a call. The call is failing and FS is generating a 488 Not Acceptable Here. After some troubleshooting, i've determined the issue is with the audio candidate list having a private IP address

    a=candidate:329485039 1 udp 2122260223 61644 typ host generation 0 network-id 7 network-cost 50

    Reading through the documentation for 3.3.X, i don't see anywhere to define a STUN server to correct this issue. 

    I then went ahead and cloned an older version of the libwebphone that used sipjs 0.7.5. I was able to define and configure a STUN server, and I was able to make and receive calls, but I had one-way audio. My computer never played audio, but my cell phone always had audio from my computer. Using wireshark, I confirmed that FS is sending the audio back to my computer, but it's like the web phone isn't getting the remote audio track or something. Have you guys run into this before? Any suggestions on resolving either issue?

    I am testing with the rc3 version of kamailio and kazoo-freeswitch 4.3.-4. Kamailio is enabled for websocket role and TLS role, and i have configured certs on it and everything. 

  5. Hey Jack,

    Welcome! I'm not sure if you've been contacted by anyone yet, but wanted to touch base to let you know that the community is thriving and if you need any additional referrals, please PM me and i'll do my best to direct you to the right resource for your needs. So... lol, i'll need an idea of what you want to do with your business and the like. Thanks!

  6. 16 hours ago, Mandeep said:


    I have been trying to configure gxp 2140 phone through provisioner but not able to do so.

    I have factory reset the device multiple times and also updated its firmware version to which is compatible with the server.

    The weird thing is that I have configured other gxp 2140 phones with same firmware version through the provisioner, they work but one of them is not working.

    Any suggestions?



    I know this sounds silly and you may have already checked a hundred times, but did you verify the MAC address of the device is what you entered in the portal?

    Can you download the provisioning file manually?

    And to confirm, you're saying that the phone doesn't ever configure itself, right? Like, when you log into the GUI for the phone, none of the SIP information is present?

  7. On 2/25/2019 at 9:53 AM, Jack Noe said:

    Hi All.

    When i want to disable missed calls when a call rings to a group, it only works if someone in the group picks up the call.

    But if no one in the group picked up the call, all phones will receive a missed call.

    Is there a way to disable the missed call completely from a ring group, so only direct calls to the device will receive missed calls ?


    We have customers that have doorbell ringing to groups, and the missed calls are annoying when no one picks up, as its not a call from a person but from the doorbell.


    Or is it possible to have a specific contact, (for example the sip username of the doorbell) have missed calls disabled complete thru the phone itself?

    The customer has Yealink T29




    Hey Jack,

    I don't believe that Yealink has this capability, and it's not really a function of Kazoo. Like, when a group is rang, all devices get the INVITE. When someone answers, they all get a CANCEL with a Reason that the call was answered elsewhere (i don't recall the specific cause code or text offhand). In this situation, Yealink says, "cool, that was answered elsewhere and not missed." When no one answers the group call, the CANCEL to all the devices contains a Reason NO_ANSWER. To the Yealink, this means that it actually missed a call and therefore presents it in the missed call list. This is a shortcoming of the device, and i'm not familiar of any device that operates the way you're requesting. 

  8. On 3/19/2019 at 11:36 AM, avig2 said:


    I have a customer who I'm moving over to 2600hz, and one of their needs is a keypad door buzzer. 

    This customer also has a few offices they rent out in their building and they would like them to be connected to the doorbuzzer system ie Algo, Cyberdata.

    Any idea how I could set this up with the other offices who aren't on 2600hz or even the same network (each tenant has their own fios or cable internet).



    You could set up the door box to have a DID. Perhaps route the DID to a menu and instead of directly to the box. Make the menu say "enter the PIN" or something. Then create an extension that routes to the door box. This way an outside account could call the DID, enter the PIN, and activate the door box. 

  9. On 9/26/2018 at 3:48 PM, mc_ said:

    @extremerotary its more along the lines of starting a dynamic conference with a custom conference profile (config for moh, alone-sound, etc) and making sure each participant is brought into the conference with the proper configs set.

    The transfer to the conference FS server should handle CCVs; are you seeing them come in as missing if the caller is moved by Kazoo to the right box?

    Yeppers. When they are transferred via Kazoo from one FS to the other, the INVITE is missing all the X-headers, and thus on CHANNEL_DESTROY, none of the important CCVs are present

  10. On 9/21/2018 at 8:46 PM, Miriam Libonati said:


    • Dynamic conferences track setup config more accurately for each participant

    Does this bullet point indicate that, when a conference is on FreeSWITCH 1 and a second call starts to process on FreeSWITCH 2 and then must be transferred from FS2 to FS1 to join the conference, that the custom channel vars are maintained and thus the channel events emitted would have the custom channel vars like owner_id?

  11. Hey guys,

    I'm just checking in to see if KAZOO-5537 is complete. In JIRA, it says the status is development complete, but I don't see any commits on the ticket (maybe because it's a private repo), but also, the expected behavior is not present in qubicle version 4.1-52. When you use the API to set the queue membership, there are no websocket events emitted. Is this improvement in version 4.2?

    Let me know, thanks!

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