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Vodia PBX

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Posts posted by Vodia PBX

  1. It is not a simple topic.

    Generally speaking, "never change a running system". This applies especially for older models where is it sometimes better to leave them running with their version, even if there are know bugs instead of taking the risk of upgrading them with new problems. 

    Apart from that, it makes sense to use the latest official releases. If you feel you need to use a different version, I would try those versions. You can do this specifically for one extension, just change the firmware link in the parameter section for the extension (in the provisioning tab).

  2. You can assign as many DID as you want to an extension. For example, you could assign one just for conferencing purposes. Depending where you are coming from, this "DID" can also contain also alphanumeric characters. So for example, you could use the alias name "conf-5434634564" as DID/alias name for the extension and then call in from a trunk that goes directly to the conference server, so that there are no carriers involved. The PBX will then take care about ringing the user up, including desktop phones, cell phones and apps.

  3. We are not against it at all, actually believe that this is a great step into reducing SPAM and increasing office productivity. There is a nice description on the Bandwidth web page (https://www.bandwidth.com/glossary/stir-shaken/) that explains the call flow. The PBX is either the "calling party" or the "called party", authentication is done through username and password. So the PBX would be the "end user"—even if the PBX is not physically on the end user premises. Do you know any SIP trunk provider that would parse STIR/SHAKEN headers?

  4. Well... Do you own that domain? If not, it would be quite a surprise if the robot would issue you the certificate! You prove that you own the DNS address by pointing it to the IP address of the PBX and make port 80 available in that address. I mean, if you try pbx.google.com it will most likely also not work unless you work there and have good relationships with the management. 

  5. We have added another option in the dropdown for the "what-to-show" option which will show the calling and the called number in the same header (in the next build). It does actually not require a code change, just another option in the dropdown, so you can also just to it by changing the template e.g. for the dom_hunt.htm web page:

                          <option value="cmc2">[[dom_acd.htm#from_cmc2]]</option>
                          <option value="$f ($t)">[[dom_acd4.htm#from]] ([[dom_acd4.htm#to]])</option>
                          <option value="original">[[dom_acd.htm#from_original]]</option>


  6. There is something keeping the PBX from picking up that call. Possible reasons are permissions (is that extension allowed to pick up the call), the call is already gone (should be easy to reproduce), that extension cannot pick up calls (e.g. because it is a hotel room extension),  or one of the extensions is actually ringing, but from a group. In the latter case there is a flag that tells the PBX weather such calls should be picked up, the default is not.

    If the star code is not handled, the PBX continues finding a way to terminate the call. Its debatable—probably its better to send the call to a IVR that tells the caller that the action failed. We'll address that in on of the next versions but here it is more cosmetic. 

  7. 488 points at a problem with the codec and/or SRTP. As a first step, I would switch to TCP to make sure it is not a SRTP problem. Did you change anything with codecs?! 

    As for the headsets they are from the PBX point of view just another DECT endpoint. You have already seen the parameter 1 trick to tell the base station what IPUI to use for what extension (actually you can also do this "manually" with the pairing process through the web interface of the M700). We did not test this here—but I would not expect any difference with another snom DECT handset.

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