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Alex Kasperavicius

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About Alex Kasperavicius

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    Advanced Member
  • Birthday 04/14/1968

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  • Gender
    Male
  • Location
    N├╝rnberg, Bayern GERMANY
  • Interests
    VOIP, telephony, cooking, drinking,
  1. HI Alex,

    Are you awake over there in Germany?

    Give me a call at my office at 909 868 5188 and my cell is 909 225 8988. I am currently on the phone with Doug Bishop - Nighthawk, and we were thinking about you! Our appendages were tingling!

  2. Yes. It isn't exactly clear is it? Here goes. To make the 870 and 820 work with PBXnSIP in its current release you must do the following: Download the three files above. Ensure they are all named properly (pnp.xml, snom870.xml and snom_3xx_fw.xml). Using SSH or some other method, put the three files you downloaded into the pbx/html directory on your PBX server or CS410 unit. Reboot the PBX Reboot your 870 phone(s) Your 870/820 should pick it up right away using PnP and you can then adjust buttons using PBXnSIP's web interface, the directory will work, etc. Note that this is a PATCH! You must remember later: When and if you decide to upgrade your PBX software to a (as yet unreleased) version that supports the 8xx series out of the box, YOU MUST REMOVE the three files you just uploaded as they will always override any new files in a new PBX version if you forget to do this. Customization Additionally, if you wish to further customize the behavior of all your 8xx phones (e.g. stuff that isn't controlled by PBXnSIP) you can log into an 8xx phone web interface, set it up the way you want and download an xml settings file with all your little customizations. Using a text editor, inspect this xml file and remove anything you don't want. Name the xml file snom_870_custom.xml or snom_820_custom.xml depending on your model and create a directory called tftp in pbx/html - so your path is pbx/html/tftp and put these xml files in it. All of your 8xx phones will use these files as the settings default so you can do something like turn off that extremely confusing "dial tone on hold feature" or make all the phones ring like crickets. But again, be careful as they will even override the settings PBXnSIP generates for each phone. Remove ANY individual phone specific settings from your custom xml files or you will have cloned phones and get very confused. If you get totally lost, just remove the files and reboot. If it's bad, wipe your phones as these settings stay there until they are specifically changed or the phone is reset. Firmware upgrade Unfortunately these phones still have some major bugs in the release software. The most annoying: No Sidetone!! (You can't hear yourself). You can download the beta version of the next update here at snom.com, but remember you may have other issues as it is still in beta. Otherwise the phones seem to work fine. I'm still testing and haven't rolled it, but it's a very cool looking phone and for many executive types that's enough. Note: I DO NOT work for PBXnSIP or Snom (but maybe should? ) these are my own observations. Use these suggestions at your own risk. Alex
  3. If you do what pbxnsip has suggested (above) you are setting up what's called a E&M (Ear & Mouth) trunk pair. I'm really disappointed that pbxnsip is offering this as the solution as it is a brutal, old-school hack. It's like someone there doesn't understand VoIP at all, or this concept of using more than one cs410 was completely off their radar when they designed it. Anyway, here's what they mean: Extension dial tone from one cs410 is an outgoing "trunk" on your windows box and vice versa. You will need to adjust dialing rules on the windows box to send all direct dial extension traffic for a specific office over the correct "trunk". So, if you have extensions in office A starting with 4xxx, and extensions in office B starting with 5xxx, make a dialing rule that that uses the correct trunk when it sees that pattern. You will also have to set this up in reverse on every system to handle calls back to the main office AND calls between branch offices. But when you're done, anyone can pick up any phone in any office and just dial the extension they want. You could do intercom and paging as well, but that requires more thought... In any case, - NOT ideal and really shouldn't be suggested or used I think - because: In this setup it uses TWO extension licenses on each system and requires an additional extension license on EVERY system for every office you add! You can't mix extensions and have to have separate number groups (e.g. 2xxx, 3xxx, and 4xxx) or have a "code" for each office (e.g. dial 7+extension for New York and 8+extension for Boston) It really doesn't scale well and can get very confusing very quickly. Ideally there should be a better way to do this using ENUM and registering each pbx with each other. I am trying to set up exactly the same thing but am facing the same problem but it seems these guys never considered this as an option. Pbxnsip hinted somewhere that this might be supported in a future release, but they haven't released a date so this might be your only (very expensive) option for now. AK
  4. I had a customer who didn't want music on hold but didn't want silence either. So I took a page from the phone company play book and created the following comfort sound: This is a standard "on hold" sound: Two bursts of 440 hz at 20 ms every 10 seconds. You can add this to your system and never have to deal with hold music issues again... AK mohbeep.wav
  5. Really? How? This isn't explained anywhere. I have both the MeetingPoint and an 870 and am manually creating an xml file by downloading the settings from from the phone and putting it in the pbx/html/tftp folder where it does get picked up - but this requires a manual xml file for each phone. How are you doing it? AK
  6. I just went through this. Get this: you have to CREATE a folder on the root. Why this is the case is a mystery and getting a clear answer on how PnP works is like pulling teeth it seems with random one sentence answers from multiple people using the same account. The jist is that the system dynamically answers PnP requests by creating answer files on the fly. If you have settings you want to change, you can create an answer file that overrides portions of the system generated one. Be careful and test, but it does work rather well once you get it down. Step One: Log in to a snom phone and configure it the way you want. From ring tone to LDAP info, put it all in and test the phone until its behavior is perfect. Step Two: Save the complete xml file off the phone onto your computer. (You have choices to save only changed values - but that didn't get everything in my experience) Step Three: Edit the xml file using a text editor and REMOVE all phone specific data like the network settings, line settings, dial plan, etc. These will still be created on the fly and if you leave them in you'll have 50 phones registering to the same account. Just jump in using any text editor. Be sure and set <settings_refresh_timer perm="">3600</settings_refresh_timer> as by default the phones never check for setting updates. This will ensure they check in. Note that you can set phone passwords in this file too. A trade off as you are exposing the passwords to anyone who would look at the answer file but easier than screwing around with PBXnSIP's HTML interface for each account. If you're behind NAT and not exposed to the internet, think about it. Step Four On your PBXnSIP box, in the 'pbx' directory CREATE a folder called 'html' and in that folder CREATE another folder called 'tftp' - so you get pbx\html\tftp Step Five Take your modified snom xml file and name it snom_360_custom.xml - It's the same process for the 300, 370, etc. Step Six Put the above file in the tftp directory you created. NOT the other tftp directory that already exists. Step Seven Power cycle another snom 360 and see if it picks up the settings. If it does, congrats! Now power cycle all your phones or your poe switch and they will all get them. If you don't do this step, the other phones won't get the new settings as the refresh timers are off. Make sense?
  7. Calling Party Control (CPC) is very standard on all modern switches worldwide so you should not have to try to detect busy signals. It wastes time (the trunk is not usable during this period) and errors can occur mid-call with conference calls, laughing, and other noises. The phone company actually drops power on the trunk for a time and that is what the CS410 should detect. Play with the detection time settings and sees if you can get it to work - it is a much better solution than using tone detection. Otherwise you may have weird problems later. Do you know what kind of switch your phone company is using? Here is a good write up on CPC and some other trouble shooting ideas: http://www.sandman.com/cpcbull.html Alex
  8. Hi All -- I have noticed that PnP no longer works when the WAN port is active. Is this by design or is this a bug? (This is with latest SW). If I were to move to TFTP only (seems like it would be easier in some respects), will pbxnsip still dynamically generate the configuration files or do I need to manually make one for each phone? Thanks, Alex
  9. I am seeing the same thing. What I've noticed is that the Cs410 stops giving answer supervision on the PSTN trunks. You make a call and the timer never starts, if you try and put the call on hold it just disconnects. Many times users are disconnected with a "timeout" error even though they are in the middle of the call. This may be the cause of the other problem where we are seeing lines simply stop responding - something is wrong with the PSTN interface. After restarting the cs410 all is well for a while. The time starts immediately when making or receiving a call using the PSTN interface. As a restart fixes this, perhaps we should put in a batch script in the interim which restarts the thing every night - crude, but it would work. AK
  10. Hi All - The snom phones have an annoying feature that plays dial tone when you put a call on hold, this can be turned off manually in the phone's web interface, but we forget to do it if a phone gets added later and is a waste of time if I have 10 phones that all need to be adjusted. There are other little phone behaviors I want to adjust as well having to do with transfer, backlight, etc. you name it. Unfortunately, it seems like various bits of the phone's configuration are sprinkled all over and it's a guessing game to see what I can and cannot control though the pbxnsip interface so I decided to look into this mysterious pnp.xml file that one can upload. It seems like one can configure a phone using tftp, pnp, and manually. However, I have noticed that if I put anything in the tftp folder, pnp just stops working forever and the server must be wiped and re-installed for it to start up again. Please, what is the BEST way to configure snom phones? I can set up a snom phone exactly the way I want it and save the settings as an xml file - but what do I do with it? The file contains phone specific information (like extension names and account passwords) that do not belong in a general configuration file. Do I just delete this and keep the bits that I need? If so, what do I do with it then? The only reference I can find on this is here and it is delightfully non specific and unclear saying only: Really? What's the "working" directory of the pbx (the path perhaps)? If I have to create an html directory, where do I get the default files to modify? What do I call these files? How does this work exactly? I am hesitant to try much, because pbxnsip has a nasty habit of turning the pnp feature off whenever it assumes manual files will be used and there is no way to turn it back on. Ever. You must re-install the pbx from scratch - which is time consuming and frustrating. -- If someone would be kind enough to step me through this: I want all snom phones to NOT play a dial tone when a call is placed on hold. How would I do this using plug n play so I don't have to log into each phone and turn this "feature" off. In an ideal world one would merely enter the the mac addresses of snom phones off the box into the account fileds on pbxnsip and the plug in the phones - and all would be configured properly. This is what I am shooting for, but I am at a loss. Any comments would be most appreciated. Thanks, AK
  11. Hi All - On a new cs410 installation trunk 1 simply busys out (short circuits or is seized) after a few hours and there is no log entry. Outgoing calls no longer use that trunk and no calls show in progress. A hard restart and all is well for now. I will see what it's like tomorrow but it's feeling like a hardware issue. Has anyone else experienced this one? This is on an AT&T 5ESS and I am getting CPC signals from the phone company. Thoughts? AK
  12. Ah, I get it - you want to set up TWO VLAN ports on the LAN side of the CS410 and make one of them for the phones and one of them for http. Nice idea, but I have a feeling this thing would get crushed with the overhead if you could set it up. If you want to use VLAN separation, you can dedicate the LAN port of the cs410 to your voice VLAN (create an untagged port on your switch and make it a member of the voice VLAN). If you also want your users on workstations (in a different VLAN) to be able to access the web interface, well - you said the cs410's WAN port is sitting on your firewall's DMZ port - just make a firewall/dns ruleset to route internal web traffic for the cs410 through to the DMZ side (and tell the cs410 to only respond to web requests from internal addresses if you're worried about security). Yeah, a bit of a pain to do but it's not that bad Adding VLAN tagging to the cs410 might be nice, but decoding tags is intensive, and you then open up a host of more configuration fun like what services are available on which VLAN, port, routing between them, and the itinerant trouble shooting, unforeseen problems, etc - and I don't think the current processor could handle the load - and I wouldn't want my VoIP processor stuck with that task, frankly. Heck, you hear a glitch in music on hold when doing a web refresh as it is. ...also, don't forget that many small/medium IT techs have little to no experience with VLANS and this is the target market for the cs410. You don't want to confuse the poor guys do you? AK
  13. You were almost right - it has to be MONO - (1 channel) - see this post for details.
  14. OK, got this to work pretty well with a few caveats. Some processing was required to the audio, but once it was done it sounded great and in theory one could set up a music server with these processed files and spit them out all day - or even subscribe to an internet radio station. In case you're not familiar with the RTP streaming format, what we're doing is using another computer to send audio packets to the CS410 on a certain port. We just tell the CS410 to listen on that port in MoH setup, set the domain preference to use that source, and we're good to go. The program I am using to generate the RTP stream is called VLC and while it does have options to transcode a file on the fly, it doesn't seem to work very well so I transcoded the audio files first using audacity This means I DID NOT make it work with standard, stereo mp3 files like we all have right away, I had to process the music files first to be: -Mono -16 bit -8 khz I also lowered the volume significantly. Fortunately, it's the same format as the moh.wav file that comes with the pbxnsip prompts, so you can use that to test and as an example of volume. If you have a bunch of music to change, you can batch script it using sox - pretty straightforward to figure out. Once you have installed VLC on a computer, use the following command line: VLC moh.wav --loop --norm-max-level=5 --sout='#transcode{acodec=ulaw,samplerate=8000,channels=1,ab=16}:rtp{dst=192.168.1.2,port-audio=4000]' We're telling VLC to loop moh.wav over and over, keep the volume reasonable, and output as ulaw, 8khz, mono, 16 bit, to a server located at 192.168.1.2 on port 4000. A full reference for these command line settings is here. Now, I realize that these settings in theory would make VLC transcode any music file, but when I tried using a stereo MP3 file it was awful. Maybe someone else can play around with different source formats. I have been playing with it for a while and pre-processing the music - while a bit of a pain - worked flawlessly. The only problem I've found is if the stream stops, the CS410 keeps repeating the last packet over and over, which sounds horrible. It would be nice if it could go silent or revert to the internal wav file if it sees the stream drop. -- So Kevin, does this get me some license brownie points?
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