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extremerotary last won the day on October 8 2019

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  1. 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.
  2. Using the DND button could work here 😉 . Otherwise, yes, you'll need to create a TOD per user and a callflow for each to toggle the TOD rules. I know of no other solution
  3. @FASTDEVICE James merged this PR for me a couple months back that was created specifically for your (and my) use-case. https://github.com/2600hz/kazoo/pull/5899 This allows you to branch to the cf_set_variable callflow module, set group_id, then go to the group. Then, in the CDRs, the custom_channel_vars will contain the variable with the value you've defined.
  4. In my experience, the callflow ID in the CDR is the first callflow that the call followed, so if you ever branched the callflow for the telephone number to another callflow ID, the CDR won't illustrate that.
  5. @avig2 Polycoms - <attendant attendant.resourceList.{x}.address="{PRESENCE_ID}" (102) attendant.resourceList.{x}.callAddress="{*0+EXTENSION_NUMBER}" (*0102) attendant.resourceList.{x}.label="{TEXT_DISPLAY_ON_PHONE}" (John Smith) attendant.resourceList.{x}.type="automata" />
  6. @mc_ That did it, thanks 🙂 For reference for the community - add this to your sipinterface_1.xml <param name="apply-candidate-acl" value="localnet.auto"/> <param name="apply-candidate-acl" value="rfc1918.auto"/>
  7. @mc_ Do you have an example of this I can test with?
  8. 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.
  9. @AlexKazoo Hey Alex, Update the system schema file. I put a PR in for this a few months ago - i'll have to check on its status... system_schemas/media
  10. 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.
  11. @esoare Yes, my testing would indicate that forwarding does not respect the ring timeout. I was hoping there was something else i could set...
  12. Hey guys, When a customer dials a callflow feature code that sets call forwarding on the user, how long is the ring timeout on that forwarded number? Is that configurable? Thanks!
  13. 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
  14. 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?
  15. @Josh Robbins Were you able to get this to work in your private cloud? If not, PM me and i'll walk you through it. But again, it's not supported by 2600, so if they do an update and break it... oh well!
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