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Vodia PBX

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Posts posted by Vodia PBX

  2. No audio at all?

    in General Setup

    Posted

    Has anyone else dealt with call drops or channel hanging in an install?

     

    (Trying to find a way back to what the problem was:)

     

    So that NAT problem is solved, now the problem is that the FXO lines hang? "Hanging" meaning that the call stays connected, although it should be disconnected?

  5. skype

    in Calling Card Setup

    Posted

    can u tell me the settins for skype as i willneed it if possible to use it as a sip server

     

    Skype does not support SIP. They are a closed user group using their own secret protocol that accesses your PC through the network, with all your data on it. Make your own judgement.

  7. moh

    in Embedded

    Posted

    can u be more explict regarding the cs410 device

     

    The CS410 is running Debian 3.1 (Linux). It is pretty much the same like a PC, but not running an Intel processor but an embededd processor type. If you feel safe dealing with the PC Debian, then it is worth trying to copy the file onto the CS410.

     

    All you need to do it drop the file in the /pbx/audio_moh folder.

  8. Call through pbx

    in General Setup

    Posted

    Is there a feature to allow you to call into the PBX say using your local home number, enter a pin code and then dial long distance to use the sip to reduce long distance fees?

     

    Yes. Either use the "Calling Card" account and call that account (your extension must have a PIN code set for that) or put your home number in as cell phone number of an extension and call the auto attendant. Then you will get a special IVR that asks you if you want to place an outbound call.

  10. SNOM 370 no DMTF and no audio

    in snom Phones

    Posted

    When I call my own extension I hear the mailbox attendant. When I push my code I hear the dtmf sounds. But it doesn't do a thing. Also I tried to enter the code and finish with #, no result.

     

    So that means the PBX is able to send media to the phone, which is good. Question is why the phone cannot send media to the PBX.

     

    How is the PBX configured? Are you using several interfaces (IP addresses)? Is it behiond NAT?

  11. Setting up trunk with Broadvox

    in Trunk Setup

    Posted

    What I would suggest in this case is to change the primary name of the domain to something useful (maybe even the IP address) and move the "localhost" to the alias name. Then the PBX will present the primary name to the outside world.

     

    But if you put the domain name into the trunk, then it should also put that name there - however that depends on the RFC3325 mode. Maybe also worth a quick try.

  12. Disable callback - just wait

    in Auto Attendant Setup

    Posted

    I turned of the offer camp, but now the caller is just disconnected, I want them to hear "All our extensions are busy at the moment, please hold the line" till the extension is available and then connect to the extension.

     

    Well, what you can do is assign a ACD for the extensions that should have this behavior. Then you get pretty much that behavior. A "one-person" ACD...

  14. Extensions on VPN

    in General Setup

    Posted

    We tried to connect additional phones via VPN from a remote location, but these phones failed to register.

     

    The network has three different subnets. There are no firewall restrictions, all ports are open. Pings, remote file access and remote desktop are all working fine.

     

    Any idea what could cause that problem?

     

    What OS? Do you see the packet coming in?

     

    What is the IP configuration?

  15. moh

    in Embedded

    Posted

    how can i upload my own wav file to this device

     

    i need to put my own music when the client callswhere do i have to go for this option

     

    You can use sftp to upload the file. For example, you can use Putty (psftp). Put it into the /pbx/audio_moh directory. Default username/password is root/root123.

  16. External Call Recording Feature

    in Call Recording

    Posted

    Can I do the same thing for system-initiated (Record all external calls) ?

     

    No. Actually, sending recordings as email scares me (even in the year 2008):

     

    One hour recording = 3600 * 13200/8 = 6 MB.

     

    Maybe you can fix the problem with a little Linux script (sorry for the meta code):

     

    while [ 1 ]; do

    sleep 10

    for i in 'ls'; do

    if [ -older than 2 minutes ] # To make sure that this file is currently not being written

    mail -s "Recording $i" <$i user@domain.com

    sleep 1

    rm $i

    fi

    done

    done

  17. No audio at all?

    in General Setup

    Posted

    comcerto:~# more /proc/net/route

    Iface Destination Gateway Flags RefCnt Use Metric Mask MTU Window IRTT

    eth0 0001A8C0 00000000 0001 0 0 0 00FFFFFF 0 0 0

    eth1 00010101 00000000 0001 0 0 0 00FFFFFF 0 0 0

    eth0 00000000 0201A8C0 0003 0 0 0 00000000 0 0 0

     

    Well, that means that the PBX has only 192.168.1.x as IP address. That is not a public IP address... If you want it in the public you have to do something about the public IP address (http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses is a good start). Do you have a public IP address? Can you use your router in router mode, no NAT? Otherwise things will get complicated or instable.

  18. External Call Recording Feature

    in Call Recording

    Posted

    We have the Call Recording License - currently set as *12 to start, and *13 to stop recording.

     

    In the Extension we use for recording - we have selected "Record outgoing calls to external numbers:" set to ON - but this does not appear to be working.

     

    What's the point in having those settings (Record calls from HUNT GROUPS, etc) - if you have to enable the *12 to start recording ?

     

    The user-initiated recordings are parallel to the system-initiated recordings.

     

    Why outbound calls are not recorded - they should. The only thing that I can think about is licsense (which seems to be okay here) and the recording location. But maybe we should just re-test here.

  19. Caller-ID Issue

    in Embedded

    Posted

    However, while I was playing with the input gains, I noticed that everytime I restarted the system the Caller-ID worked for the first call. After that, it seems that almost every call is missing the Caller-ID.

     

    Do you think there may be some other issue going on?

     

    The only thing that rings a bell is that the polarity change detection is off in the beginning (by default). This is because the gateway wants to "learn" if it is available or not. Maybe on next reboot try to turn it off and see if that makes a difference.

  21. Crackling and Popping noises

    in Embedded

    Posted

    We have replaced our allworx-6x with the CS410.

    Once we went to 2.1.6.2448 on the black box, things have been much more stable.

     

    But, we have an intermittent noise problem.

    It happens both on the one analog POTS line we have, as well as the SIP trunks with Callcentric.

    We are using Snom-360 and linksys SPA942...both phones seem to have the noise but the linksys is worse.

    If I transfer the call from the Linksys to the Snom, the popping will also transfer.

     

    It is a loud Crackling, Popping sound.

    It is consistent during a single call, but may disappear on the next call.

    Any ideas?

     

    The gain adjustment was a little bit buggy on the 2.1 version. There is a new version http://www.pbxnsip.com/cs410/update-2914.tgz that should be better with the gain.

     

    If you want to be able to move back to the old state, just make a backup of the /pbx directory before doing the upgrade.

  24. Help needed

    in Fax Setup

    Posted

    Does PBXnSIP support NSE switching to T.38? I tried using both NSE and re-invite on the sipura but neither seemed to work. In re-invite mode I could see the sipura send the invite with the T.38 stuff, but even then it didn't work...

     

    Well, the PBX supports a "pass-through" through the PBX - that includes the SDP. If the Sipura sends a Re-INVITE, you should see another Re-INVITE going out on the other side of the call.

     

    More investigation needed methinks, it's just a pain that the customer is miles away and isn't keen on us having remote access...

     

    Understandable. Maybe you have the chance to reproduce it in the lab.

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