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Vodia PBX

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Posts posted by Vodia PBX

  4. Security

    in General Setup

    Posted

    If you're storing a hash (like a simple MD5) of the user's web password and PIN, I wouldn't expect the hash to break if the alias or domain name is changed so long as you don't use the alias or domain name to calculate the has.

     

    Well the problem is when the user decides to change the username you cannot generate a new hash. If we are using the hash method we have to request users to re-enter their passwords.

  8. SPA 962 Dial Plan

    in Linksys Phones

    Posted

    Has anyone found a good dial plan to use with the SPA962 phone? right now when i dial *67 it doesnt block the caller ID, it just says enter number: and when i do so the caller ID is still not blocked

    but on a GrandStream phone wheni dial *67 it says : yoru caller id will be blocked on outgoing calls

     

    it seems that teh linksys phone has its own sets of codes that will cause it not to send the correct code to the PBXnSIP system?

     

    That is a kind of Linksys classical problem. The default dial plan of the phone makes it hard to dial star codes on the PBX. See http://wiki.pbxnsip.com/index.php/Linksys for some dial plan examples.

  10. Email address question

    in Email Topics

    Posted

    When creating accounts as an extension, can multiple email accounts be listed in the Email Address: Field, such as accountname@domainname.org; accountname2@domainname.org; accountname3@domainname.org; etc.

     

    The logs of my email server show error messages that state "<<accountname@domainname.org>... Unbalanced '<' (hold)

     

    I understand that the email program is adding the < but is the Unbalanced error due to the multiple email addresses in the field? If so, is there a work around (other than creating mailing lists)?

     

    Thanks in advance.

     

    Ehh. Looks like a bug to me. Workaround could be to use the form "<accountname@domainname.org>; <accountname2@domainname.org>".

  11. Enable 911

    in ITSP's

    Posted

    here is the log on the system whent hey try to call us:

     

    Very good. Maybe you increase the log level and look for messages like "Trunk xxx sends call to xxx" (log level 5)? Then we can see what the PBX tries to do with the INVITE.

  12. Stress Testing

    in General Setup

    Posted

    I should add that later on i get this message from pbxnsip:

     

    The call from sip:99@xxxxxxx.com:5060 to

    sip:sipp@xxxxxxx.com:5069 has been disconnected because of media timeout (120 seconds), 0/6000 packets have been received/sent

     

    i feel like im missing something in my scenario to disconneced the rtp stream correctly.

     

    Maybe you have to turn "strict RTP" routing on. I believe that test tools don't have to be very NAT-friendly, so the PBX gets one-way audio.

  13. SPA525 G

    in Linksys Phones

    Posted

    "The SPA525G ... such as the Cisco SPA9000 Voice System or a Broadsoft or Asterisk system" from their user manual PDF, is their any way we can make that certain?

     

    Well, at least Broadsoft is using SIP, and I guess with Asterisk they are also using SIP so I would say it works. But only the real device will tell how much exactly works.

  14. Security

    in General Setup

    Posted

    I've noticed that all the extension passwords (website, SIP, and PINs) are stored in clear text on the pbxnsip server. I'd like to suggest that you store a hash of the website and PIN passwords. I know the SIP password has to be able to be transmitted to the phones. It could use a reversible encryption.

     

    We thought about that also. However, when you change a domain name or change one of the alias names than a hash would not work any more. Encrypting it with a hardcoded key only "obscures" the passwords (until someone gets the bit secret out of the code). Encrypting it using the private key of the PBX (used for TLS) would be a possibility.

     

    At least the sys admin login uses a hash for the password!

     

    And of course, file system access should be strict. This is not a public area.

  15. 3.1.1

    in New Features/Versions

    Posted

    Here's what's happening...

    - I have "Username/password required" selected for TFTP -> Generate passwords.

    - Under 3.0.1.3023, I can go to http://<my server's IP>/provisioning/polycom_phone_<my phone's MAC>.cfg and it'll prompt me to authenticate. I put in my extension number (only one domain) and password and it displays the XML configuration file no problem.

    - Replace 3.0.1.3023 with 3.1.1.3118 and try the same thing and you won't get authenticated.

     

    Because the Polycom's can't get their configuration when they're booting up, they try to reboot hoping this will cure the condition and the endless cycle begins.

     

    If I change TFTP -> Generate passwords to anything other than "Username/password required", it works ok. Of course, I want username/password required.

     

    Well, you have to set up the the password on the Polycom phone (see http://wiki.pbxnsip.com/index.php/Polycom). Did you do that?

  16. Pressing 5 (with time/date envelope) when listening to VM causes CS410 to Crash

    in Embedded

    Posted

    No eth2:

     

    #This file was automatically generated by the IP PBX.

    auto lo

    iface lo inet loopback

     

    auto eth0

    iface eth0 inet static

    address 70.x.x.x

    network 70.x.x.x

    netmask 255.255.255.192

    broadcast 70.x.x.x

    gateway 70.x.x.x

     

    auto eth1

    iface eth1 inet static

    address 1.1.1.1

    network 1.1.1.0

    netmask 255.255.255.0

    broadcast 1.1.1.255

     

    Log File shows:

     

     

    Logfile

    Clear or Reload the log.

    [1] 2009/01/05 16:31:15: Starting up version 2.1.0.2115

    [8] 2009/01/05 16:31:15: Route: eth0 462a4400 ffffffc0

    [8] 2009/01/05 16:31:15: Route: eth1 01010100 ffffff00

    [8] 2009/01/05 16:31:15: Default Route uses 70.x.x.x

    [7] 2009/01/05 16:31:15: Found time zones AKDT AKST PDT PST MDT MST CDT CST2 EDT EST ADT AST NDT NST BST CET CST CAT IST GMT

    [1] 2009/01/05 16:31:15: Working Directory is /pbx

    [5] 2009/01/05 16:31:17: Starting threads

    [7] 2009/01/05 16:31:17: UDP: Opening socket

    [0] 2009/01/05 16:31:17: UDP: bind() to port 5060 failed

    [0] 2009/01/05 16:31:17: FATAL: Could not open UDP port 5060 for SIP

    [7] 2009/01/05 16:31:17: Opening TCP socket on port 5060

    [7] 2009/01/05 16:31:17: Opening TCP socket on port 5061

     

    Not sure why???

     

    Oh there is probably an old PBX running still (don't reboot the box until the problem is solved!). You can find the process with netstat (e.g. netstat -anp|grep 5060). Kill it.

  17. Call Forking Options

    in Dial Plan Setup

    Posted

    Is there a way in the dialplan that you can write a rule that only applies to a call that is forking out to a call phone? I want to be able to send the outbound forked call out a specific trunk (since only some of our trunks will allow us to send any caller ID information).

     

    You mean that cell phone forks should use a different dial plan that regular calls or other call redirections for that extension? What is the purpose? Caller-ID presentation?

  18. Scheduled Conference - How TO

    in Conference Room Setup

    Posted

    I have read the WIKI and the manual on scheduling a conference with PBXNSIP, but I must be missing something. I have changed the conference mode to "scheduled" conference" but I don't see where I can send out emails from the PBX with the conference PIN code or schedule the times/dates of the conference.

     

    I'm running 3.0.1.3023 (Darwin) on a Mac Mini using Mac OSX. I've checked every screen, and I can't seem to find where to do this. I've attached a screen shot of my conference extension for reference. Thanks in advance for the help.

     

    You see this screen only if you log in as a user. The domain or sys admin cannot set up conferences.

  19. Counterpath Soft Phone - Domain Address Book?

    in General Setup

    Posted

    I use the Eye Beam Counterpath soft phone with PBXNSIP version 3.0.1.3023. I'd like the ability to have the soft phone pull the domain address book from the PBX. I've read some documentation from the soft phone and it says that is supports something like this, but VOIP is not my strong suit, so here I am. I have copied the part of the manual below from the soft phone. Thanks in advance for the help.

     

    Whow, we don't support that yet but it looks interesting. Do you have a pointer to the documentation?

  22. 1G servers

    in General Setup

    Posted

    I have a 3U blade server with like 20 1Ghz servers. wondering how many calls 1GHZ would typically handle.

     

    Maybe is would be less extensions per server but more dense in overal extension (3U)

     

    That is really hard to say. But lets assume 50 calls that you be 1000 calls for the whole box. Nice!

  23. Transferring Directly to VM

    in Polycom Phones

    Posted

    i have another idea...please advise if this would work.

     

    basically currently the issue is that when the user enters 8121 the phone does not recognise 1 as a prefix to an extension and therefore cuts off the fourth number because the dialplan states that 8 is a prefix to an internal extension and therefore the phone is saying send this call to 812.

     

    however if we took 8 out of the dial plan (there are currently no extensions beginning with 8) then maybe the phone would wait for the fourth digit (in this case 8121) and when the receptionist hits # it would work?

     

    for testing purposes can you please tell me how i can go about removing the 8 i(ie the dialplan should state [2-7]??

     

    The [2-9]11 is essentially for 911. Maybe you just write 911; maybe also include 411 if someone wants to dial it. This way you can take the 811 out.

     

    In the end, you need to test test test. IMHO life would be easier if they can move the 1xx extensions out of the way or just accept that they have to dial 9 for an outbound call (to "seize" a line, haha). If you have to tailor them their own dialplan that will mean some work.

     

    Or they get over the whole thing and press Dial after entering a number. Thanks to the several billion cell phones out there that is not such an crazy idea!

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