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  1. If everything is in the LAN, all you have to do is to put the multicast paging group on the button of a phone and provision the PA1 and the phone. Then when you press the button it will be sent from the phone to the multicast group and make the PA1 play it back. The PBX will be unaware about it—no CDR and also no call recording.
  2. One important question is if this happens within the same LAN. If the answer is yes, then multicast can be a good option (unless call recordings is required). In that case, you can just provision the PA1 be part of the paging group and have the phones permission to page that group. Then the PBX will take care about provision everything to work peer to peer. If your desktop phone does not support sending multicast RTP or you want to use the feature e.g. from a PSTN or Microsoft Teams account, then the PBX can still generate multicast. This would require that the PBX and the PA1 are in the same LAN. If neither a phone that can generate the multicast RTP nor the PBX is in the LAN, then we are talking about sending the RTP over unicast SIP. This is easy if there is just one PA1. If there are many PA1, it becomes more and more difficult to have them all play back the page as the load on the system goes up, which is especially a burden if you are on hosted PBX with limited bandwidth into the LAN. In that case there is an option in the PA1 to relay the unicast RTP into multicast RTP. However the PBX does not support provisioning that kind if scenario and this would have to be set up manually.
  3. Wir arbeiten an einer Lösung die auch mit der Deutschen Telekom (und sicher auch noch anderen Anbietern) sauber funktioniert. Das wird noch ein paar Tage dauern.
  4. Try to scan the QR code with something else (e.g. Lens). Does the content make sense, especially the address of the PBX? Would the cell phone be able to reach the address?
  5. You should open ports 443 (HTTP TLS) and if you want to use LetsEncrypt certificates, you must also open port 80 (cannot be another port). For UDP, you need to make sure that the PBX UDP ports (by default 49152-65535) are also forwarded to the PBX.
  6. "http:" is not an action—this would be configured on the ActionURL page...
  7. We need the domain name somewhere otherwise it would not work in a multi domain environment. Though I agree this should be done on HTTP level (hostname). Anyway we are looking into this.
  8. Yes—certificates work only with DNS addresses. snom does this for a long time. This is in /reg_pnp_settings.htm setting "Use domain name instead of IP address".
  9. Beim Thema SRTP schien es etwas hin- und herzugehen. Zwischendurch ging es nur wenn man SRTP explizit ausgeschaltet hatte. Jetzt scheint es so zu sein, dass SRTP/SDES doch wieder geht (vermutlich gab es zu viele Geräte die es so machen so dass DTLS in der Praxis zu viele Probleme bereitet hat). Wir haben die Vorlage wieder entsprechend angepaßt. Thema Codec ist auch schwierig. Das liegt weniger an der Telekom, sondern daran dass die das SDP weiter reichen an die andere Stelle, die dann teilweise mit nicht G.711 ins Schlingern kommen. Daher ist G.711 sozusagen der kleinste gemeinsamer Nenner mit dem es am wenigsten Ärger gibt. Wenn G.722 nicht nativ auf den Endgeräten unterstützt wird klingt es schlechter als G.711. Bezüglich Zuordnung eingehender Anrufe sollte IMHO praktisch immer ein Präfix verwendet werden. Dann braucht man keine Telefonnummern anlegen, sondern nur dafür sorgen dass die Nummer hinter dem Präfix einer Nebenstelle (kann auch eine Gruppe sein) entspricht.
  10. The app uses web socket—so there will be no UDP. Depending on how you are logged in, the PBX will generate a HTTP or HTTPS URL for you.
  11. Practically everything today uses SNI—because practically all web traffic is on multi-home web servers. There are unfortunately still some SIP phones out there that did not have SNI turned on. In a nutshell, there is no way to use them with TLS in a secure (validate certificate) way if there are more than one tenant on the hosted PBX. The only way to get this working is to certificate validation off.
  12. We also have a OpenWRT build, see the "OpenWRT (MIPS)" target in http://portal.vodia.com/downloads/pbx/version-65.0.xml for example it runs on https://openwrt.org/toh/hwdata/zbt/zbt_wg3526_16m
  13. Why using such an old version? We are now at 65.0.
  14. We have a template for the 1620—might be much easier to use the template than going through all the settings and o this yourself. You only have to set the provisioning server to the address of the PBX, add the MAC address and start the provisioning process.
  15. Usually you need to use https—though e.g. in Safari you can tell the browser than http is fine (mostly for testing). However there should be some kind of warning if https is not allowing WebRTC to start.
  16. We had already a similar problem problem when it came to the question what outbound ANI to use for an outbound call. The rule was to use the one from the last ACD the user logged into. It makes sense to not only associate the ANI with the ACD, but also assign the call the ACD. We'll do that in the next version 65.0.5. Maybe we can make that even graphically visible in the soft phone.
  17. Looks like this is an area that needs to be revisited. Especially the possibility to send out weekly and monthly emails should be very useful in many environments.
  18. The dial plan pattern normally does not use the +-form. It uses the form in which a human would read the number—for NANPA this is just the 10-digit number and for international numbers, its 011xxx. (2-9) is not a valid pattern. Do you mean [0-9]? BTW if the pattern (priority number 30) matches the plan will stop there. That means you don't need to make that exception in the line 40. You can probably just use * for "everything else". Or you could use the pattern [2-9]xxxxxxxxx.
  19. I think the only place where you will have to change anything is on your SIP trunk header. This is not a problem of the dial plan. E.g. there is a setting in the SIP trunk that tells the PBX to present the number in E164 format (which is without the leading +).
  20. The number of SMS providers will probably for us mushroom like the SIP trunk providers—which is a good thing! So lets try to address this in a way we can maintain that for a long time. We'll add a provider "custom" that just collects the username and password (which is stored encrypted in the filesystem so there is no need to have it in clear text lying around in the file system). Then you can drop a JavaScript into the PBX that will do whatever the provider needs to send and receive messages. We'll provide an example on doc.vodia.com on how to use this, maybe just using the pushbullet.com sample.
  21. Yes this setting was made for soft phones that sometimes literally dial what users entered, for example "(617) 399 8147" which should be "6173998147". Especially when dialing from the address book or call history.
  22. Well, there is the possibility to edit a post! 1+ is very confusing indeed. Anyway there is another post about it, in a nutshell I would just look at the logs along the processing of the call and make adjustments so that the numbers make sense in each step.
  23. We should look on the process on how the dialed number gets from the original caller to the trunk. When the PBX feeds the number into the dial plan, it formats the number (by default) into the "human readable" country-code dependent format. In the US that would be 01144xxx. Then the replacement comes up with the destination number, which is then fed into to SIP trunk which may reformat the number once again. Keeping this in mind my approach would be to turn logging on and look at the numbers as they go through this process and then make adjustments to the dial plan and the presentation in the trunk.
  24. Outbound is relatively easy—inbound is the problem (but that's where things get interesting). If you have a provider that works we'll be happy to add them to the drop down. Maybe we should also start adding providers that support only outbound SMS.
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