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Everything posted by Kristan

  1. Kristan

    CDR to mySQL

    I've followed the wiki guide, but my pbxnsip box doesn't seem to be interested in logging anything to mySQL. There's nothing in the logs at all regarding CDR and I'm a bit unsure as to where the CDR URL should go. I don't have an admin->settings->general->CDR URL field? I've tried setting it in the CDR filename field - is that right? Running
  2. Hi all, Not really a PBXnSIP question, but I was hoping someone here might have some tips. I have a PBX customer who has moved to Ireland, and can only get satellite broadband. We've got a phone up and running, but the quality is "iffy" to say the least. There's echo, delay and starting the conversation means waiting 10 seconds or so before the audio sorts itself out (I guess as the phone/PBX figure out the latency and jitter). Is there anything that can be done either at the PBX side or at the phone side (linksys SPA-942)? I was thinking of trying something other than G711 also maybe? Anu suggestions would be appreciated! Thanks, Kristan
  3. So as it stands, PBXnSIP publishes the state of a monitored phone by sending the call details to the phone doing the watching, and it's up to the phone if it should translate this into BLF for lamps etc. and/or display it on the screen? It's caught me out a few times today already, I've picked up the phone to answer a call that's for someone else and been very confused by the dialtone! Regarding amending the polycom provisioning file to stop this behaviour, it'd be nice if there was an easy way to edit the copy the PBX has internally (or download a copy), rather than us having to ask support for the current files - would that be something you'd consider? It'd make my life easier!
  4. Just upgraded a few of our phones here and discovered this new "feature". To my way of thinking, this should be difference between "Watch the calls of the following extensions" (I can see who is ringing who) and "Watch the presence of the following extensions" (my sidecar & buttons should show the BLF status)? However, it seems to get the buttons to work, I have to "Watch the calls of the following extensions", and then I also get to see who is ringing who. Not what I want. The presence option doesn't seem to do anything. Any ideas?
  5. Hi All, As an internal SLA requirement, I need to be able to get some statistics from PBXnSIP on how many calls where answered, how long the average call was ringing for before getting answered and any calls not answered within 10 seconds. I can get all of this from parsing the email that sends out the queue status each night but I was wondering if there's a better way to get at the stats, or if I can change the email so it's in a bit more of a machine readable format (XML attachment?). In the past I've done this by logging all the calls to a database by reading the files in the CDR folder each night, but it seems a bit overkill when all the information I need is mailed to me each night. Oh, as an afterthought - is there any chance of you guys adding the ability to send the CDR records to a database at some point in the future? Like mySQL, or MS-SQL? That'd be a very nice feature, and would make writing a CDR tool very easy Thanks, Kristan
  6. Solved! Ancient firmware version
  7. I've installed OCS 2007 R2, and have a mediation server. I've followed the OCS setup guide, and can make calls from the Tanjay phones we've got to PSTN (via PBXnSIP), and incoming calls come up on my PC with the Office Communicator Client. Presence between the phone and desktop seems to work, but I cannot get the Tanjay to ring when a call comes in - it only gives me the option to answer on the PC (which isn't very helpful, as I don't have a headset ). Calls between Tanjays work... Have I missed something fundamental here??
  8. http://technet.microsoft.com/en-gb/office/bb735838.aspx Is there any particular reason pbxnsip isn't listed on here? Or is that something you guys are working towards?
  9. The firmware will be the same version, as we don't use the individual sip.ld's, we use a single large sip.ld with all the separate phone versions in - phones provision a bit slower initially, but it's easier to maintain. We'll move the phone to the main office and try it there - I would expect it to work over TFTP, but can't see why HTTP would work any differently depending on where it is? And if we provision with TFTP, we'll have to swap it back to HTTP when it goes back out to the remote site, and the same thing will happen again?
  10. Anyone? Customer is getting a bit annoyed now at not being able to us their conference phone, when previously they were! Thanks
  11. Hi guys, Upgraded a customer to the latest and greatest and using the new PnP stuff, had a few issues with the phones. Got all these sorted now, but having a problem with the SoundStation IP 4000 conference unit. Basically, the setup is : Head office, phones using TFTP and MAC addresses to auth (trusted option ticked) Branch office, over VPN, phones using HTTP with usernames/passwords set on phones This works, however when it comes to the conference spider, setup with HTTP at the branch office, it hangs on boot. Looking at the wireshark trace, I can see the following : GET /2201-06642-001.bootrom.ld to which the PBX replies with a 302, redirecting the to login page. Which the phone asks for, to which the PBX replies with the 302 to the login page. Which the phone asks for... and so on, and the phone hangs. The phones also do this, but instead of a 302, get a 404 (correctly). The only difference I can see is in the GET from the conference unit, it passes it's user agent as FileTransport PolycomSoundStationIP-SPIP_4000-UA/ whereas the phones send FileTransport PolycomSoundPointIP-SPIP_650-UA/ Is the PBX seeing the SoundStation bit and not recognising it as a provisioning request and therefore redirecting to the login page? Is this something we can solve by adding a line to the pnp.xml (does it still exist for the new PnP stuff?) or is there something else amiss? I have the full pcaps if you want to have a look. Thanks
  12. I noticed the "templates" section on the website that we can download a template for. I'd say 90% of our installs the same trunk setup, same dial plans, domain and system settings etc.. It'd be nice (and would make our job easier) if we could just install a PBX and upload something to it and have all that setup for us and we just have to enter the extension numbers and names Is that something we could build ourselves too?
  13. They need to be separate unfortunately! Is there any way to do this, or do I need to tell them just to check one of the phones?
  14. Hi All, I have a customer with an office 10 with a single hunt group with all the phones in, and they just want any voicemails to go to a single mailbox - that's fine, I can send it to one of their extensions as the final stage in the hunt group. However, they want to get notifications on voicemails on all the extensions rather than just the one I select. Is there any way to do this, or a recommended way? The phones are a mix of polycoms and siemens C475IP gigasets. I'm not sure if the gigasets support remote VM - there seems to be an option but I've not had a chance to have a play yet... Thanks, Kristan
  15. Bit of a feature request rather than a question I suspect, but here goes. A lot of our PBX's are sold to people who have several phones on a single extension, i.e. one in the office, one at home, one at their villa in France. A common complaint is that if they're on the phone in one location, and a call comes in to their extension, instead of getting a busy and going to VM, it rings the other extensions. To me, that's obvious, and the solution is to put DND on the other phones when they're not there. However, they're human, so sometimes forget. Is there a solution to this? Something like an extension level setting that says "single user" or something, so if the PBX gets a busy on one phone, it knows not to carry on ringing it elsewhere? I know it's a bit of a bodge, and I don't really have a good solution. However I can see what the user wants, and it'd be nice if I were able to make him happy Any ideas??
  16. Still the same, but the invite is just xxx@pbxnsip.wm.local If no-one has a general "how-to" guide for configuring this with PBXnSIP don't worry too much, I'll keep playing and see if I can get it working.
  17. Agreed - it still dialled ok though! I think the @ symbols was my fault, I had it setup to add @pbxnsip.wm.local when it dialled, and it was doing it each time I tried to resend.
  18. Should also say I've had this working in the same config with a sipura 3102, so I know T.38 and faxes work, just need to figure out what I've done wrong with this one
  19. Hi guys, Just wondering if anyone has a standard setup for running this with PBXnSIP? I've got it registered to a PBX extension, but it doesn't seem to try to negotiate T.38 as part of the invite, it just sends it as ulaw, and the fax fails (it dials answers/correctly etc.). Also the invite seems to have the domain name appended rather a lot. Any ideas before I get onto their support guys? : INVITE sip:xxxx@pbxnsip.wm.local@pbxnsip.wm.local@pbxnsip.wm.local SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bK1106 From: IPFax <sip:xxxx@pbxnsip.wm.local@>;tag=IPF_PORT_0001_1105 To: <sip:xxxxxx@pbxnsip.wm.local@pbxnsip.wm.local@pbxnsip.wm.local> Call-ID: f0627200-e41a-44eb-9328-b7152a8149ff@ CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:6192@pbxnsip.wm.local@> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Authorization: Digest username="xxxx",realm="pbxnsip.wm.local@",nonce="2101959bb16d637f6f099251b5ddc659",opaque="",uri="sip:xxxxxx@pbxnsip.wm.local@pbxnsip.wm.local@pbxnsip.wm.local",response="8ddf7a06f7ca94c7d46d228767c2873c" Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=IPFax 0 0 IN IP4 s=SIP Fax Call i=IPFax c=IN IP4 t=0 0 m=audio 49154 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv
  20. Hi, We've got a customer asking about this, so now I'm asking you guys The basically want to pick up the phone, enter a client code, then dial the number, and have the PBX record this somehow. Is that how it works? What does that end up in the CDR as? Ta
  21. Is there a list of major features that are in V3 that aren't in V2? Just trying to put together a mailer to convince our customers to pay us to upgrade them
  22. As requested - these also include some features for the 3.1 version of the polycom firmware - older versions should just ignore them and carry on. Standard disclaimer that these work fine on my phones and PBX, they may not on yours polycom_phone.xml polycom_sip.xml
  23. What firmware version are on you with the polycoms? You can download the latest here, and just put the sip.ld and bootrom.ld files in the tftp folder of PBXnSIP http://www.polycom.com/emea/en/support/voi...oint_ip430.html 3.0.3 rev B seems to fine for me. Be aware though that you'll need to update your polycom_sip.xml and polycom_phone.xml files if you've crafted them by hand. I'm happy to share mine, but they include a couple of bits specific to us which you might want to change.
  24. Anyone installed this and tried the setting in PBXnSIP for "Watch the calls of the following extensions"? It seems to break the phone display a bit - the BLF lights seem to get stuck flashing in some kind of "pickup" mode when they take/make a call, and the main screen displays the calls that have been taken/made by other extensions, and there's no way to clear them. Any ideas? Running PBXnSIP 3023 btw, with my own hacked together version of the pbxnsip reference files. I can happily supply examples if you want
  25. Can I grab the latest and greatest to give this a go then? Also on a similar note, I've just updated my phone to the latest polycom firmware (3.1.0B) on 3022 and now when a call comes into the hunt group and it rings on colleagues phones (which I'm monitoring via sidecar) I get the option to pick it up on my phone? It doesn't seem to clear properly though - must be some strange bug in the polycom firmware methinks..
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