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cwernstedt

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About cwernstedt

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  1. Is there a way to get automatic blacklisting to block all attempts from an offender IP address instead of blocking IP+originating port as separate instances? As it is now it appears that, say, 46.166.151.43:5137 and 46.166.151.43:5151 are blocked as separate entities, so if 46.166.151.43 tries with thousands of other additional originating ports, the black list will be filled with thousands of entries. (Plus my email address being hammered with thousands of notifications...)
  2. Is there a way to periodically and automatically change the participant access code (PIN) of a conference using a script that accesses the API or manipulates settings files? It's an "ad-hoc" conference that users regularly access. I have studied the API documentation, but I can't find the name of the attribute to change, and it's unclear if a new access code can be submitted via the API at all. /CW
  3. OK. I think to move things forward a bit faster than trial and error, I should pay for some consulting from someone skilled with setting up auto-provisioning with Snom and Vodia. (Should be fairly standard scenario.) To make this happen, do you recommend that I purchase some premium support tickets?
  4. I don't PnP the phone yet. It's not in the same LAN as the PBX so not sure how to do it properly at this point, but I'd like to solve this when the installation grows. Is PnP necessary for as-feature-event subscription?
  5. Is DND state syncing between snom D785 and Vodia 60.0.2 supposed to work? using_server_managed_dnd1=on ...but state changes on the PBX don't show up on the phone.
  6. @Support I sent the numbers in a private message to you here on the forum.
  7. I couldn’t find any SIP messages going out to the trunk when the particular cell phone number is used. SIP messages were sent when the working number was used.
  8. I'm trying out the 60.0.1 release. Having some trouble: - Cell pone redirect for an extension (114) to a certain number (+6421NNNNNN) stopped working. - Calling +6421NNNNNN from the 114 extension works. - Cell pone redirect for the extension works when set to a different but similar number (+6427NNNNNNN) . Note that the number that works has more digits than the number that doesn't work. It is not likely that the issue is with the outgoing trunk, because the log doesn't indicate any INVITE messages sent through the trunk when 114 is called and cell phone is set to +6421NNNNNN , and the trunk operator (Twilio) doesn't log any attempts or errors. Dial Plan used by 114: 101;Twilio (CH Office);;00*;"sip:+\1@\r;user=phone";;false 103;Twilio (CH Office);;0*;"sip:+41\1@\r;user=phone";;false104;Twilio (CH Office);;5500*;"sip:+\1@\r;user=phone";;false105;Twilio (CH Office);;550*;"sip:+41\1@\r;user=phone";;false106;Twilio (CH Office);;+*;"sip:+\1@\r;user=phone";;false The setup works in v59.0
  9. cwernstedt

    Let's Encrypt SSL Certificate support?

    +1
  10. cwernstedt

    caller id passthrough

    One of our operators, Twilio, says it's OK with our PBX "spoofing" the CID when calling out on their trunks for as long as the CID belongs to someone calling into their system. This would enable us to forward the actual originating CID to our cell phones. Caller (CID#1) --> Twilio Trunk --> PBX --> Twilio Trunk -> Cell phone (sees CID#1) However a caveat is that CIDs from calls originating from non Twilio trunks as well as from internal extensions should not be spoofed (or can't be spoofed). I'm uncertain about how to configure trunks properly for this scenario. Any ideas?
  11. cwernstedt

    Calls Routing/Redirection

    I have a similar problem to the one discussed above. I need to differentiate between calls coming in on two trunks to the same provider (Twilio). One trunk as defined on the Twilio side enforces TLS at extra cost, but I don't want all incoming calls from Twilio to go via this trunk. Is there a way that I can make the PBX route Invites like the two below to different trunks? Invite version 1 (I want this to go to the first trunk): Invite version 2 (I want this to go to the second trunk)
  12. cwernstedt

    One way (or no) sound for clients connecting with VPN

    As it turned out, I had forgotten to put in all of the local networks in the IP Routing list under SIP settings. Like this: (covers all private address spaces) 10.0.0.0/255.0.0.0/[PBX private IP address] 172.16.0.0/255.240.0.0/[PBX private IP address] 192.168.0.0/255.255.0.0/[PBX private IP address] 0.0.0.0/0.0.0.0/[PBX public IP address] The amazing and puzzling thing is that things worked at all from misc networks in the 10.0.0.0- series before this change.
  13. Hi, My customer switched from an old pbxnsip installation to a new Vodia PBX (latest version) . On the old installation, SIP clients running on phones/laptops connecting with VPN (IPSEC or OpenVPN/SSL) to the PBX's location had no issues with placing calls through the PBX. On the new installation, on the other hand, there's one way audio (IPSEC) or no audio (OpenVPN) . I suspect it has to do with NAT (rather than with VPN as such) because SIP clients on various LANs that are interconnected via IPSEC tunnels have no issues. Any suggestions? Note again: Things worked straight out of the box with the old pbxnsip, but it doesn't work on the new Vodia installation. Cheers, Christian
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