Jump to content
Vodia PBX forum

Vodia PBX

Administrators
  • Content Count

    9,725
  • Joined

  • Last visited

Community Reputation

0 Neutral

About Vodia PBX

  • Rank
    Advanced Member

Profile Information

  • Gender
    Male

Recent Profile Visitors

192,415 profile views
  1. In the PBX, paging accounts can be in unicast or multicast mode. The latter one would be a "Multicast Paging Group". Entering 224.1.1.2:4000 there is a good choice. You can provision the button also from the PBX without having to touch the phone—the PBX would then set up the VPK button automatically. Just select the paging account for the button and hit the save button. As for the echo, that is a good point. There must be a setting that tells the phone that it should not play back its own RTP stream. IMHO it would be reasonable that this is "on" by default. Unfortunately Grandstreams many models all have different provisioning profiles which make it hard to take care about each and every model. Other vendors keep the name of the settings across large amount of devices.
  2. 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.
  3. 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.
  4. 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.
  5. 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?
  6. 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.
  7. "http:" is not an action—this would be configured on the ActionURL page...
  8. 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.
  9. 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".
  10. 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.
  11. 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.
  12. 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.
  13. 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
  14. Why using such an old version? We are now at 65.0.
  15. 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.
×
×
  • Create New...