Jump to content

Kristan

Members
  • Posts

    211
  • Joined

  • Last visited

Posts posted by Kristan

  1. According to pbxnsip that should be on the fly... When you press the Addressbook button, it dynamically downloads the address book.

     

    Hmmm, only the first 9 in this case! Maybe I need to latest and greatest version, I think they're a couple of revisions behind..

  2. Just noticed that the 300's that have been auto provisioned on a new system have their "buttons" (mute, directory, redial etc) aren't working anymore, presumably as they're set to "buttons".

     

    How do I stop the PBX doing that?

     

    ta

  3. That was my thinking too. I think the we'll jus have to specify the extension and password for each manually and again manually create it a config file - it's going to be a pain if they want to change extension numbers though...

     

    I understand you guys are working pretty closely with Polycom now - I don't know if you maybe have access to resources we don't as mere resellers? :lol:

  4. Ehh - somehow the provisioning file got lost (we don't even find it any more...). Would be great if you can give us access to the file that you have :lol: ...

     

    You can download them from here :

     

    http://downloads.polycom.com/voice/netlink...02_v130_005.zip

     

    Basically to set them up you do the following :

     

    1. Configure SSID, WEP/WPA/WPA2 key and SIP username and password on the handset

    2. Configure the DHCP boot server option to point to PBXnSIP (option 66 - same as normal polycoms)

    3. Boot phone, it will get DHCP (assuming your wireless bit is correct), go to your TFTP server and ask for :

    • slnk_cfg.cfg - this points it to the software bin files
    • whatever.bin - it checks it's got the current version of whatever files are specified above
    • sip_allusers.cfg - general settings - codec, sip server IP, DTMF mode etc.
    • sip_extn.cfg - where extn is the SIP username specified manually on the handset in step 1.

    I asked Polycom if it were possible for the phone to request a file by MAC etc. like the soundpoint series do, but it seems that it can't.

    So I don't think there's much scope for autoprovisioning the phones really, as you have to hand make the sip_exten.cfg file for each extension.

     

    Unless you guys know another way? :lol:

  5. No, it's just that the UPDATE implementation of most devices is actually broken and we had the same effect as you - until we turned the support of UPDATE off.

     

    And yes, if the FXS detects FAX it should send a Re-INVITE with T.38 in it. If it does not then that is a sign that the T.38 is not enabled on the device.

     

    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...

     

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

  6. Do you see anything with UPDATE? That was a common problem with some devices that indicate they support it, but when you send UPDATE nothing happens.

     

    Otherwise, Wireshark is most useful in these cases... Maybe you can send my a PM and we'll take a look at this.

     

    Hmmm, I don't see an updates from anywhere at all. I assume I should see this at the point the sipura detects the fax and should send an UPDATE to the PBX to change negotiate T.38?

     

    Thanks for your help.

  7. Hi All,

     

    Could do with a bit of help getting faxes and T.38 working with sipura 3102's and PBXnSIP. Setup is :

     

    Vegastream PRI -> PBXnSIP -> Sipura 3102 -> Analouge fax

     

    Inbound faxes seem to work fine (I can't see anything T.38 related in the initial invite, but I assume it does an NSE or something afterwards). Outbound is where the problems are. The sipura can either do T.38 via NSE or a re-invite, both seem to follow the same course :

     

    1. Initial invite, PBX sends call to the vega, everyone ACK's and OK's and the call begins.

    2a. NSE mode, nothing else happens (RTP is flowing both directions) then about 30 seconds later the sipura sends a BYE and tears the call down.

    2b. RE-INVITE mode, sipura detects fax, does a RE-INVITE with the T.38 bits on the INVITE and same thing happens, nothing for 30 seconds, BYE, call ends.

     

    I've got full logs from the PBX and wireshark trace, I just want to see if anyone has this working? I suspect it's settings on the sipura I need to change somewhere, but the fax options look pretty straight forward and are all set as I'd expect them to be.

     

    Help <_<

  8. For example - If John transfers a call to Jane and while Jane’s phone is ringing, and John hits TRANSFER again - the call will likely fail.

     

    Wow, I've never seen that in any of our Polycom systems. I've just tried it now and if you press transfer again before the other party picks up it just carries on ringing. This is on v3 of the polycom firmware though - any reason why you don't want to upgrade?

  9. Ok -I've downloaded and installed the PAC but I guess the agent group support is something that needs to be added. The customer is happy to beta test this - do you have an ETA on when it would be available? There are only 5 agents so a projector is a bit over kill :)

  10. Hi All,

     

    We've setup an agent group for a customer and they would now like for the agents in the queue to be able to access the status of the queue they are in. Unfortunately there doesn't seem to be a way to do this without giving them domain admin rights.

     

    Is it possbile to maybe have an agent login (the extension number) or have it in the "lists" section of the extensions options?

     

    Or is there another way I can get deployed quickly for them?

     

    Thanks

  11. I just don't think I've got the command line right - it doesn't seem to be propper G.711..

     

    I'm using :

     

    vlc -vvv file.mp3 --sout "#transcode{acodec=ulaw,samplerate=8000}:rtp{dst=10.50.4.190,port=4000}"

     

    I can't see it sending anything in wireshark now I've looked, so I'm definitely doing something wrong...

  12. Has anyone had this working?

     

    I've had a play with VLC, but all I get is screeching when I put a call on hold, I suspect I've not got the codec/mux settings correct, but I can't see what they should be set to. Does anyone have definite working settings?

     

    ta

  13. Hi All,

     

    We have a customer who is having trouble with their conference room. They have an inbound VoIP number routed to them which is aliased to a conference room. When there's more than a couple of participants, audio drops out every 3/4 of a second or so, it's really regular. The PC it's on is only running at 2-3% CPU, and 4-5 8K streams shouldn't saturate a 2MB SDSL pipe.. Any ideas or possible things to look at?

     

    Thanks.

     

    Kristan

  14. What experience created thr practice of both IP and LocalHost?

     

    The snoms used to drop registrations when watching other extensions if it didn't use the hostname. Localhost from memory just means the PBX will accept all connections, and the IP just in case the DNS isn't setup or available.

  15. Hi guys,

     

    I thought I'd post up the installation procedures for our generic polycom installs to help people starting out for the first time, but also so anyone comment on good/bad practice and what they think?

     

    Obviously this is for the UK and biased towards our way of doing things, but any criticism is appreciated!

    • On DNS server, setup A record for PBX
    • On DHCP server, set option 66 to IP of PBX
    • Download and install latest PBXnSIP build and UK prompts (unzip to audio_en folder and overwrite)
    • Delete noise.wav and create a 0k file called noise.wav
    • Login and under admin settings, change:
      • System name
      • Timezone to London
      • Admin password
      • Process Affinity mask to 1 (I know you don't need to, but I like to)

      [*]Set license

      [*]Under logging, set:

      • Log level to 7
      • Log length to 500
      • SIP Logging: other messages Yes

      [*]Edit domain settings and set primary name to DNS name, aliases to localhost and IP

      [*]Enter domain admin and setup trunks (in our case a SIP Registration for SIP and a SIP Gateway for the ISDN gateway)

      [*]Setup dialplan to route operator and emergency calls to ISDN, all other calls to SIP

      [*]Create extensions/hunt groups etc. as customer requires. Make sure extensions are set to permanent assignment

      [*]Edit domain settings and set default dialplan to the one you created

      [*]Download latest polycom firmware and bootloader and place in TFTP folder (along with associated files)

      [*]Place polycom config files in html folder (these are our customised ones - you can skip this if you want standard pbxnsip/polycom ones)

      [*]Plug a phone in, set the provisioning mode to TFTP and sit back and make sure it updates and registers correctly

      [*]Test outbound calls route correctly and inbound ones present caller ID / called ID correctly

      [*]Job done!

  16. Hi All,

     

    Just want to double check what I have in mind will work.

     

    Vega400 PRI gateway with 30 channels and 50 DDI numbers (supports T38). Customer wants some of those to be faxes, so in PBXnSIP I setup the number as a normal extension. I've got sipura 3102s (which also support T38). These will register to the extension, and faxes should magically work? :lol:

     

    Thanks!

×
×
  • Create New...