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  2. Hi, thank you for your indication. About page (reg_pnpparm.htm) and the field "Snom General" What happens if these parameters are already present in the general parameters. I'll give you an example. I really don't like the call transfer method used in the vodia template. I prefer a more traditional method with simultaneous blind + attended transfer. This should also be the method most appreciated by users (judging by the requests). Using these parameters it is possible to obtain a completely automated transfer that allows you to transfer the call in blind mode (transferring and hanging up) or to transfer the call in attended mode, talk to the colleague and then complete the transfer by hanging up. There is no simpler way than this. To do this, just modify these parameters in all Snom: __________________________ <disable_blind_transfer perm = "PERMISSIONFLAG"> on </disable_blind_transfer> <transfer_on_hangup perm = "on"> VALIDVALUE </transfer_on_hangup> <transfer_on_hangup_non_pots perm = "on"> VALIDVALUE </transfer_on_hangup_non_pots> ____________________________ Some of these parameters are already present in the template of course. By placing them on the page (do m_ext3.htm) in the "Snom general" field does this result in an override of any parameters present in the template? At the moment I have directly modified the template of the single phone but, since parameters are generalized for all snom models, it would be preferable to do it according to your advice. The doubt arises from the fact of not creating conflicts. Finally, I would like to ask you for clarification on the placement of template.xml files within the vodia folder structure. I am not referring to the files generated by the parser but to the original files. Where are the various .xml files related to the various templates placed? I couldn't find them in the folders. Are they present in the system database or cannot be modified? I am used to working intensively with other systems on the modification and customization of the templates. I've been dealing with it for many years and I'm very familiar with all the major brands. I was therefore wondering if it was possible to duplicate some templates and customize them or even insert the templates for the missing phones or adjust those that present problems or are not fully functional. I thank you in advance.
  3. Hi, thank you for your indication. About page (reg_pnpparm.htm) and the field "Snom General" What happens if these parameters are already present in the general parameters. I'll give you an example. I really don't like the call transfer method used in the vodia template. I prefer a more traditional method with simultaneous blind + attended transfer. This should also be the method most appreciated by users (judging by the requests). Using these parameters it is possible to obtain a completely automated transfer that allows you to transfer the call in blind mode (transferring and hanging up) or to transfer the call in attended mode, talk to the colleague and then complete the transfer by hanging up. There is no simpler way than this. To do this, just modify these parameters in all Snom: __________________________ <disable_blind_transfer perm = "PERMISSIONFLAG"> on </disable_blind_transfer> <transfer_on_hangup perm = "on"> VALIDVALUE </transfer_on_hangup> <transfer_on_hangup_non_pots perm = "on"> VALIDVALUE </transfer_on_hangup_non_pots> ____________________________ Some of these parameters are already present in the template of course. By placing them on the page (do m_ext3.htm) in the "Snom general" field does this result in an override of any parameters present in the template? At the moment I have directly modified the template of the single phone but, since parameters are generalized for all snom models, it would be preferable to do it according to your advice. The doubt arises from the fact of not creating conflicts. Finally, I would like to ask you for clarification on the placement of template.xml files within the vodia folder structure. I am not referring to the files generated by the parser but to the original files. Where are the various .xml files related to the various templates placed? I couldn't find them in the folders. Are they present in the system database or cannot be modified? I am used to working intensively with other systems on the modification and customization of the templates. I've been dealing with it for many years and I'm very familiar with all the major brands. I was therefore wondering if it was possible to duplicate some templates and customize them or even insert the templates for the missing phones or adjust those that present problems or are not fully functional. I thank you in advance.
  4. Yea... another common problem was that the previous default number of connections was just 50, which is very low. You should set that to 500 or even more (/reg_ports.htm). As for the login we have a secondary login page "rawlogin.htm" - you are not the first with that problem!
  5. Yesterday
  6. Not quite fixed after all. It works for a little while then it renders pages with huge logos and unusual layout, then after trying ctrl-F5 (firefox's shortcut to *fully* reload) it stops responding. I'm seeing these lines at the logs: [6] 20190525003418: Last message repeated 3 times [3] 20190525003418: Current number of requests 50 has reached maximum 50, connections not accepted [6] 20190525003418: 140 more requests pending to acme-v02.api.letsencrypt.org:443 [6] 20190525003419: Last message repeated 2 times [3] 20190525003419: Current number of requests 50 has reached maximum 50, connections not accepted [6] 20190525003420: 140 more requests pending to acme-v02.api.letsencrypt.org:443 [6] 20190525003422: Last message repeated 5 times [3] 20190525003422: Current number of requests 50 has reached maximum 50, connections not accepted [6] 20190525003422: 140 more requests pending to acme-v02.api.letsencrypt.org:443
  7. FIXED! removed webUI customization (pbxwebai and webpages directory) and it works.
  8. OS is linux/CentOS. There is not firewall (wget ...localhost...). The ports _are_ open (notice that wget reports that it gets connected but I also verified with ss). rawlogin has exactly the same behavior (I get connected but get zero bytes from the PBX). I rebooted (it's deep in the night here) with no success.
  9. Hi, Please find the link here: https://doc.vodia.com/starcode_anonymous_calls
  10. After you change the domain provisioning password, you need to re-provision the phones. You can do this through the web interface, in the extension list.
  11. nothing to do, the advice you gave me does not work, now with the pswd inserted within the phone via web but as a user and not as an admin, so you can not do anything
  12. you want to tell me that if I enter in the fields "Authentication password" and repeat a pswd I want (example 34567@#) for all extensions will be accessed via web with admin/34567@# ?
  13. If you want to log in to the web interface of the phone, you need to set and use the domain provisioning password (/dom_settings.htm). By default, this is a random password - yes it is not easy to guess
  14. Alright we'll fix this. Thanks
  15. I wanted to report a bug in the starcode link indicated in: /dom_feature_codes.htm https://doc.vodia.com/starcodes_anonymouscalls
  16. I'll correct myself, The provisioning also following the online doc works partially, you must enter via web in the Grandstream phone and press the button "Start" to do the provisioning from Vodia. So you have to launch the command from the phone, but it becomes impossible because you can not enter via the web on the phone as it seems that it is impossible to trace the pswd (so I was told by you in another post).
  17. OK but you need to intervene manually via the web, enter the phone and make the appropriate customizations, as I am for the templates are not complete, if it is not possible after provisioning to make changes via the web in the phone everything becomes impractical. So I need to understand if it is possible to somehow enter via the web in the phone or it is impossible. I am testing the server, I am an installer and the solution of an inaccessible phone I do not think it is acceptable for any installer and no situation. So my question is the same, how do I access a phone via the web after provisioning?
  18. Looks to me like the PBX could not open the sockets for HTTP/HTTPS. Maybe the firewall does not like the new executable? What OS is that? Check with netstat is the ports were opened.
  19. Every phone has a different place to enter the SIP password for an account. Generally I would recommend to use the automatic provisioning for VoIP phones. There are many other settings that should be set, it is very difficult to do that manually. And the PBX is automatically assigning a password to the VoIP phone which is different from the SIP password (every extension has many passwords).
  20. Can you try with http://localhost/rawlogin
  21. I understand that it is essential to activate the DMZ etc. ... on the firewall with the IP address of the server Vodia. I'll try and update you tnx Damiano
  22. Hi, Maybe this will help you understnad better: https://doc.vodia.com/admin_sip_and_audio Look (Ctrl + F) for the word "SIP IP Replacement List" on here.
  23. sincerely I don't understand where to insert my public ip in the parameters of the Vodia, can you kindly indicate me the path where to insert my public ip (example 33.33.33.33) and where to insert the local ip range (in my case 192.168.1.0/24)? Thank you in advance, Damiano
  24. You don't need STUN. The SBC does take care about this. Please see https://doc.vodia.com/server_behind_nat for some basic information about how to run a server behind NAT - this is called "near end" SBC.
  25. I'm sorry, but I can't get my public ip on Vodia. I did the NAT of the RTP range and ports 5060 and 5061 on my firewall to the local ip of the Vodia but the remote phones (with STUN active) are recorded but there is no audio. Probably I miss to insert Public IP on Vodia and what else? my local range is 192.168.1.1/24 and Public IP is 33.33.33.33 (example) where I enter these values correctly, I looked at the documents online but honestly I did not understand much
  26. Had no problem in test server but after upgrading our production system to 63.0 we lost access to the web interface! Tried from various IPs, from same LAN and from localhost with no success. Ports 8080 and 443 are open and LISTENing, we get connected but then the pbxctrl gives zero bytes back. I'm providing the ouput of tests from localhost with wget and openssl below. Please help as soon as possible. wget -vvvv http://localhost:8080/ --2019-05-24 13:40:19-- http://localhost:8080/ Resolving localhost (localhost)... ::1, ::1, 127.0.0.1 Connecting to localhost (localhost)|::1|:8080... connected. HTTP request sent, awaiting response... No data received. Retrying. [...same error again and again...] wget -vvvv https://localhost --2019-05-24 13:28:21-- https://localhost/ Resolving localhost (localhost)... ::1, ::1, 127.0.0.1 Connecting to localhost (localhost)|::1|:443... connected. Unable to establish SSL connection. openssl s_client -debug -host localhost -port 443 CONNECTED(00000003) write to 0x21d3500 [0x21d3580] (289 bytes => 289 (0x121)) 0000 - 16 03 01 01 1c 01 00 01-18 03 03 58 2a 59 35 05 ...........X*Y5. 0010 - 31 9e c9 3a 58 b9 82 80-ad 03 9c ee cf 4b 2a 1a 1..:X........K*. 0020 - ed 50 8c 11 cf 2b 4a 98-bd be 24 00 00 ac c0 30 .P...+J...$....0 0030 - c0 2c c0 28 c0 24 c0 14-c0 0a 00 a5 00 a3 00 a1 .,.(.$.......... 0040 - 00 9f 00 6b 00 6a 00 69-00 68 00 39 00 38 00 37 ...k.j.i.h.9.8.7 0050 - 00 36 00 88 00 87 00 86-00 85 c0 32 c0 2e c0 2a .6.........2...* 0060 - c0 26 c0 0f c0 05 00 9d-00 3d 00 35 00 84 c0 2f .&.......=.5.../ 0070 - c0 2b c0 27 c0 23 c0 13-c0 09 00 a4 00 a2 00 a0 .+.'.#.......... 0080 - 00 9e 00 67 00 40 00 3f-00 3e 00 33 00 32 00 31 ...g.@.?.>.3.2.1 0090 - 00 30 00 9a 00 99 00 98-00 97 00 45 00 44 00 43 .0.........E.D.C 00a0 - 00 42 c0 31 c0 2d c0 29-c0 25 c0 0e c0 04 00 9c .B.1.-.).%...... 00b0 - 00 3c 00 2f 00 96 00 41-c0 12 c0 08 00 16 00 13 .<./...A........ 00c0 - 00 10 00 0d c0 0d c0 03-00 0a 00 07 c0 11 c0 07 ................ 00d0 - c0 0c c0 02 00 05 00 04-00 ff 01 00 00 43 00 0b .............C.. 00e0 - 00 04 03 00 01 02 00 0a-00 0a 00 08 00 17 00 19 ................ 00f0 - 00 18 00 16 00 23 00 00-00 0d 00 20 00 1e 06 01 .....#..... .... 0100 - 06 02 06 03 05 01 05 02-05 03 04 01 04 02 04 03 ................ 0110 - 03 01 03 02 03 03 02 01-02 02 02 03 00 0f 00 01 ................ 0120 - 01 . read from 0x21d3500 [0x21d8ae0] (7 bytes => 0 (0x0)) 140427521087376:error:140790E5:SSL routines:ssl23_write:ssl handshake failure:s23_lib.c:177: --- no peer certificate available --- No client certificate CA names sent --- SSL handshake has read 0 bytes and written 289 bytes --- New, (NONE), Cipher is (NONE) Secure Renegotiation IS NOT supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : 0000 Session-ID: Session-ID-ctx: Master-Key: Key-Arg : None Krb5 Principal: None PSK identity: None PSK identity hint: None Start Time: 1558694617 Timeout : 300 (sec) Verify return code: 0 (ok) ---
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