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Everything posted by extremerotary

  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!
  16. Clumsy, but you could register line 1 to user A, and line 2 to user B, then log in/out on the correct line. I know, I know, terrible workaround, but functional
  17. What firmware version are you running on it? Anything prior to 3.3 does not have the provisioning server info available in the GUI. You can check by (on the phone) going to menu > status > application (doing this from memory, so i may be a bit off). I believe these phones support at least 4.0, so you should be able to upgrade the firmware to achieve this, but it's not necessary if you can physically reach the phone.
  18. Hey Mattia, There are a lot of conference options in the works! Currently, the main functionality comes from the APIs. They give a wide range of moderator options via a programmed UI to do things within conferences. We've had very good success with using them ourselves in our UI. One obstacle we are facing is, in a clustered federated environment is that during a nightmare transfer of a caller to a conference running on a different freeswitch server, some channel vars are not passed to the receiving freeswitch server, so the conference event emitted is missing some data (such as owner_id for an on-nat call). We are currently working with 2600 and the freeswitch team to resolve that. But yeah, the APIs are where the real controls come into play!
  19. Looks great guys! I also understand there are some performance enhancements in 4.2, so don't forget to showcase them! As part of the qubicle updates, will there be an event emitted when an agent gets on and off a non-queue call since qubicle is now aware of non-queue calls? "Added ability to load all queues and recipients with single API call" - What is this new API call? Any update on wiretap with the queues? As part of the call recording storage updates: "Improvements to Call Recording storage - Ensure storing call recordings to an HTTP URL works as expected (Google Drive / AWS)" Does that indicate that these storage options are supported in 4.2?
  20. So you mean, if nothing is in there, notifications are disabled, but if I put some bogus address in there I, as the sender, will get the notification (in addition to the bogus email)?! That would work!
  21. @Josh Robbins Hey Josh, There is an unsupported module for eavesdrop (written by a third-party) that works okkkkay...It only does eavesdrop, and not whisper or barge. In the hosted platform, there are significant updates required to give/restrict this person from listening to that person, etc. So the business logistics behind it require some deep thought and forward planning + development. I wonder if the community would be interested in funding 2600 to make it their own and incorporate it into their code base and hosted platform. Unfortunately it's one of those things that customers ask for, and will never use, and so it's never gotten much traction. I've only ever confirmed that it works to "listen in", and it's current state is rudimentary and needs quite a bit of love, for it's rather easy to circumvent the "permissions" required to validate the caller is allowed to listen to the other user.
  22. Hey guys, When an email-to-fax is sent using faxboxes, is it possible to configure the notification to only be sent to the email address that sent the fax? For example, if the smpt_permission_list was "@2600hz.com" and James sent an outbound fax, if the notifications.outbound.email.send_to is set to fax@2600hz.com, then everyone in the distribution list would have a copy of the fax attachment, and know that James sent out a mass advertisement fax about pivot. I would like it if only james@2600hz.com received the PDF attachment and notification that the fax was successful (or failed). Also, 2 side questions: 1. Can the notification email be configured to not attach the PDF (using notify, not teletype)? 2. What is the "media" element on the faxboxes document used for?
  23. Hey guys, Does anyone out there have a setup guide for configuring storage to use Google Drive? I read a very good article on the doc site regarding storage plans for S3, but I'm looking to use Google Drive. I understand the basics, but I'm cloudy on exactly how to configure this. For example, from the doc site I see the schema for the storage attachment google_drive. I get everything except "settings.oauth_doc_id". The description is "Doc ID in the system 'auth' database". Does anyone have a guide on how to create this doc and how to setup the oauth in google for this purpose? Thanks!
  24. @denverUser Hey, can you provide the log lines above and below the error from kamailio, and answer a few questions? What version of kazoo, ecallmgr, freeswitch, and kamailio are you running? I understand that it's a registration question with Kamailio, but all the versions together will help to isolate if it's a versioning issue between anything. You can use 'rpm -qa | grep {{ something }}' to find out the installed versions (where {{ something }} == "kazoo" for example). Then, if running version 4.0+, from the command line, you can run 'kazoo-applications status' to see what version(s) are actually running (as opposed to installed). This output will also help in general to see what apps you have running, etc. Is this a single-server install? I'm sure you've triple-checked the SIP creds, but I have to ask! Feel free to remove any personal info from the logs, like IPs or realm names and I'll take a look!
  25. @Karl Stallknecht Let's see if we can get a quick response from the team. I'd be interested in setting a "media_id" on a conference doc, but it doesn't look like it functions that way on their doc site.
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