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  2. OK, but isn't that what is shown in the video that was done on July 9th? That the name changes as hot desking information is used? So what am I not getting or is it something that is not working properly and needs to be programmed?
  3. No those screens are not the point - the PBX provisions the "Programmable Keys". I have just checked here a GXP 1628 where the display comes from the "Account Name", which does not change with the hot desking. Looks like it would be worth changing that as well with the hot desking, there should be no negative side effect with other devices.
  4. Great for this Tips G:M, i will try that out tonite when slow use of it. REALLY APPRECIATED IT !! I LOVE VODIA MORE, with each minute that pass ! -M
  5. Yes no problem the PBX is a great soft switch . From the 3CX system is just like any other VoIP extension - it has to register and it has to answer challenges for outbound calls. In the log above the "Received incoming call without trunk information and user has not been found" is the problem. It is okay that the trunk was not found (this is not a trunk from the PBX point of view) but the user must be found. Try to use a name like "trunk123" in the account name, then this should be working with no problems. There is no need to terminate calls for remote systems - for the PBX this is a simple call from an extension to a trunk.
  6. Today
  7. They did that, trying with 1 and without 1 to dial when we conducted tests.. and endedup the same.. I havent try yet: the setting in Trunk where say: Trunk may terminate calls for remote systems -dont know if this move will cause any difference. Hope this helps.
  8. Hi. Understood. Let me explaing a little better as i can. I been using Vodia-PBX as a small Softswitch. I have couple of UCMs (GS-PBX) attached(Registered) to VodiaPBX as extensions. They work fine. But i have a customer that have several offices that uses 3CX-PBX for years, and would like to join our services (due to reduce their current monthly bill), but, its been hard to make their 3CXPBX to sucessfule dial outbound from their far end extensions attached to the 3CXPBX. It works fine backwards, their extension get the call from outside. So this is basically what is going on. Theyre just making a new trunk out as they do to attach them selves to any other provider, but as soon trying to achieve the same with us, it gives that error. I will go ahead and reply them with your last comments about it, and expect a feed back soon. Meanwhile, is this can help you understand what im trying to achieve, glad it will English not my first lang. Thanks again.. very much.
  9. I am sorry, I am not sure I know what you are asking. I set the line to use extension 9998. And It shows "General Extension 1" on the phone. It is set that way so that it can be used without someone logging on the phone as this phone (and several others) will be in an area where workers will come from other offices and hot desk in to make calls as their extension. However, when a visiting worker is not there, the phone needs to be able to dial out as a general number. As per the video in this thread, I had thought that when someone hot desks in, the extension would change from "General Extension 1" to "User A" or whatever their name is. I auto provisioned the phone, and when the phone got it's information, it populated the "Account Name" to "General Extension 1".
  10. Hmm - you did not set a label for the private line? What do you see on the Grandstream display? The PBX uses the "label" to control what you see on the screen, hopefully the Grandstream show the label content and not some Grandstream - controlled content. There are so many models it gets sometimes hard to have them all work the same way...
  11. On TCP, Grandstream GXP1628 works well with one of our live phones here. Rest is up-to you. The models you have and what firmware you want them to have is your choice completely and doesn't depend on the PBX.
  12. Ok, so what is the latest firmware for the GXP21xx phones that is non-issue causing? Thanks.
  13. Also make sure you're on the latest and the non-issue causing firmware of the GXP phone. That can also be one of the causes.
  14. Yes, used that link. Confirmed that the PBX updated to 63.0.3. Rebooted several phones. Still the same.
  15. They were automatically provisioned. And yes, that is how we did it for Yealink phones and the settings come down. Grandstream... nope.
  16. Please update using this link: http://portal.vodia.com/downloads/pbx/version-63.0.3.xml Have you checked after clearing your cache on your cell phone browser. Also try from Chrome browser on the cell phone with incognito window.
  17. Did you automatically provision the phone? Then the LDAP settings should be set up for you and everything should be working already. (Like this does for the Yealink)
  18. If you register as extension, the "From" header has to use the exact account number. The PBX does not perform any re-formatting of numbers - if you enter 6173998147 for example, it will reformat it internally as +16173998147. So you should try to use that +-number or just use a number like 435s2w38 which cannot be misinterpreted as DID.
  19. You can see the version number on the status web page - make sure it is 63.0.3. If the display does not update immediately, does it update after you restart the device?
  20. Interesting - the phone should not freeze... IMHO you should also inform Grandstream about this, with a PCAP for the LDAP port. At first glance the LDAP should work fine, it does work fine with a lot of other phone models...
  21. I just installed last versione of vodia (63.0.2) and some gxp1610/15 grandstream phones (fw and When I click on LDAP search softkey button and I search for like letter "A", phone freezes and I've to reboot disconnecting the power. Someone can reproduce this issue with a grandstream phone? Thank you. Not sure is a bug on gxp fw or other thing. Regards, Alessandro Marzini
  22. You've used this link right? http://portal.vodia.com/downloads/pbx/version-63.0.3.xml
  23. Yesterday
  24. Hi. This is the version im currently using since April i think. Software-Version: 63.0.1 (Win64) But i noticed that there is the same version in the maintenance but the date is different as 7-12-19 (63.0) A little confuse here Thank You For issue 2: I explain again, i think i wasnt clear enought. I changed the logo for ours. But the logo doesnt show in mobile version in the WEB GUI as in the Desktop version. It doesnt get replaced. Hope this helps. -M
  25. Also depends on your version. Make sure you're on 63.1 (it's in beta version but is stable). A screenshot for comparison will be better to understand.
  26. Hi. I have customers with 3CX-PBX registered to a Vodia-PBX Extension. When an extension, registered under 3CX-PBX dial out to any PSTN or CELLphone, following the route to achieve this, which is : 3CX-PBX Extension to Vodia-PBX Dial Plan, the call just give busy tone or message that cannot go thru. We have been trying to figure it out why is doing such. Calls from PSTN/CELLphones thru Vodia-PBX Extension to 3CX-PBX extension, does work fine, and reach the end 3CX-PBX using VODIA as front. Here im including logs FOR when the call wont go out,, and Vodia-PBX seems somehow unauthorizing this calls. (Perhaps, im missing something) LOGS: [5] 23:35:33.404 PACK: SIP Rx xxx.xxx.xxx.37:5060: INVITE sip:787xxxxxxx@vodiaserver:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.37:5060;branch=z9hG4bK-524287-1---cb4f5e5ecd96ff77;rport Max-Forwards: 70 Contact: <sip:1787xxxxx00@xxx.xxx.xxx.37:5060> To: <sip:787xxxxxxx@vodiaserver:5060> From: "1787xxxxx00"<sip:1787xxxxx00@vodiaserver:5060>;tag=ff841e65 Call-ID: 7D7XJvdt0rso98dxPHpvpg.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem (1581) Remote-Party-ID: "1787xxxxx00"<sip:1787xxxxx00@vodiaserver:5060>;party=calling Content-Length: 288 v=0 o=3cxPS 10777037333594112 20496333810958337 IN IP4 xxx.xxx.xxx.37 s=3cxPS Audio call c=IN IP4 xxx.xxx.xxx.37 t=0 0 m=audio 9342 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv [6] 23:35:33.404 MEDI: Port 126: Allocating port for SIP Call-ID 7D7XJvdt0rso98dxPHpvpg.. [7] 23:35:33.405 MEDI: 126: SRTP tx keys: ixj/5ztTLQVlip9UIa6crgu/4ZmDoo2TtkJY7Mjm 9C36CB0E [8] 23:35:33.405 TRUN: Could not find a trunk (1 trunks) [9] 23:35:33.405 SIP: Resolve 44968: aaaa udp xxx.xxx.xxx.37 5060 [9] 23:35:33.405 SIP: Resolve 44968: a udp xxx.xxx.xxx.37 5060 [9] 23:35:33.405 SIP: Resolve 44968: udp xxx.xxx.xxx.37 5060 [9] 23:35:33.406 GENE: UDP (IPv4): Opening socket on [9] 23:35:33.406 GENE: UDP (IPv4): Opening socket on [9] 23:35:33.406 GENE: UDP (IPv6): Opening socket on [::]:16168 [9] 23:35:33.406 GENE: UDP (IPv6): Opening socket on [::]:16169 [7] 23:35:33.406 MEDI: Port 126: Allocated ports 16168 and 16169 [5] 23:35:33.406 GENE: Received incoming call without trunk information and user has not been found [5] 23:35:33.406 PACK: SIP Tx xxx.xxx.xxx.37:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.37:5060;branch=z9hG4bK-524287-1---cb4f5e5ecd96ff77;rport=5060 From: "1787xxxxx00" <sip:1787xxxxx00@vodiaserver:5060>;tag=ff841e65 To: <sip:787xxxxxxx@vodiaserver:5060>;tag=17b336e9f3 Call-ID: 7D7XJvdt0rso98dxPHpvpg.. CSeq: 1 INVITE Content-Length: 0 [9] 23:35:33.406 SIP: Resolve 44969: aaaa udp xxx.xxx.xxx.37 5060 [9] 23:35:33.406 SIP: Resolve 44969: a udp xxx.xxx.xxx.37 5060 [9] 23:35:33.406 SIP: Resolve 44969: udp xxx.xxx.xxx.37 5060 [5] 23:35:33.406 PACK: SIP Tx xxx.xxx.xxx.37:5060: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP xxx.xxx.xxx.37:5060;branch=z9hG4bK-524287-1---cb4f5e5ecd96ff77;rport=5060 From: "1787xxxxx00" <sip:1787xxxxx00@vodiaserver:5060>;tag=ff841e65 To: <sip:787xxxxxxx@vodiaserver:5060>;tag=mrikppojw1 Call-ID: 7D7XJvdt0rso98dxPHpvpg.. CSeq: 1 INVITE User-Agent: Vodia-PBX/63.0.1 WWW-Authenticate: Digest realm="",nonce="d115aea06ce7ae170b63f81578054797",domain="sip:787xxxxxxx@vodiaserver:5060",algorithm=MD5 Content-Length: 0 [5] 23:35:33.407 PACK: SIP Rx xxx.xxx.xxx.37:5060: ACK sip:787xxxxxxx@vodiaserver:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.37:5060;branch=z9hG4bK-524287-1---cb4f5e5ecd96ff77;rport Max-Forwards: 70 To: <sip:787xxxxxxx@vodiaserver:5060>;tag=mrikppojw1 From: "1787xxxxx00"<sip:1787xxxxx00@vodiaserver:5060>;tag=ff841e65 Call-ID: 7D7XJvdt0rso98dxPHpvpg.. CSeq: 1 ACK Content-Length: 0 Hope this helps. -M
  27. Hi. ISSUE 1: I have noticed that, when using SSL to open the WEB GUI from a remote address, i cannot see the list under Currently Active Calls. Call Logs list is displayed or shown fine. 'ISSUE' 2: When using the WEB GUI in a mobile, the image loaded to be shown in the GUI is never shown as in a desktop is. Maybe it is the way is been set, but it will be nice to have it also be shown under mobile scenario too, so for example, when showing Customers or when customers Picking ") Hope this helps. -M
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