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Alex Kasperavicius

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Everything posted by Alex Kasperavicius

  1. 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
  2. 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
  3. 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
  4. 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
  5. 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?
  6. 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
  7. 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
  8. 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
  9. 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
  10. 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
  11. 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
  12. You were almost right - it has to be MONO - (1 channel) - see this post for details.
  13. 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=,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 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?
  14. OK, this took a little digging - for anyone else lost, go to: System --> Settings --> Configuration A special upload field for ringtones.xml is at the very bottom of the page. ...and you can upload the above attached ringtones.xml file AK
  15. I have found the PnP functionality to be VERY touchy - especially if you are playing around with it. If I put ANYTHING in the tftp folder and a phone downloads it, it seems to break the PnP function forever. I mean, you can't bring it back unless you back up the server, clear the tftp folder, clear the 'generated folder' (off the pbx root) , factory reset the server and restore from the back up. Really, it blows. I may be wrong, but from my fumblings with the things, (both a windows version and a cs410) it consistently breaks in this way. You get TFTP or pnp, but not both and you can't switch back and forth. In any case, for automatic phone setup here are your choices: - Manual registration: A true pain as you must log into each phone to set it up . Not recommended unless you have an un-supported phone (like the snom Meeting Point). - TFTP (Option 66) The older way where you must manually create whatever load files needed for your phones and drop them into the tftp folder. When you boot a phone, it gets the tftp IP address from your DHCP server and pulls the files from the tftp server in pbxnsip (or, potentially any TFTP server you have). - PnP This is the newest (from what I've read here) way and you need to merely plug in a supported phone. It will do a broadcast discovery and find your pbxnsip server (make sure you have only one with pnp turned on!) and the server will magically auto generate the settings from the choices you've made in your pbx setup and send them to the phone. The tftp folder IS NOT USED - it stays empty. It will also check the software version of your phone and automatically upgrade it (based on the URL string you have in the configuration screen). In your accounts section, if you have each phone's mac address in the registration tab, you can assign extensions. If you put a * in the mac address field, new phones will randomly register with the available extensions and the mac address will populate. You can then copy and paste the mac addresses where you want them, re-boot the phones (or power cycle your poe switch) and the extensions will be correctly assigned. Again, I have noticed that it can be really, really touchy and even putting some settings back will not get it working again the way you might expect it to. You CANNOT use both tftp and pnp on the same pbxnsip server. If you use tftp even once, pnp will break forever and you must factory reset the server. Set up a test machine and do a fresh install (or just back up your settings and do a factory reset as above) and you can see how it works. Once it all clicks in your head and you understand it, then give it a whirl. Hopefully with the 4.0 release the boys in Deutschland will give us options to better control this behavior, but it seems like they just wanted to get the functionality working and kind of crammed it in the latest release. You don't get a whole lot of feedback and the above information I learned though painful trial and error.
  16. This misspelling persists in all versions of pbxnsip. Because I noticed it again in the beta preview for 4.0 I figured I'd bring it up: The default names for most of the extensions are mis-spelled as "fourty" - which displays on the phones. This is embarrassing if a customer sees it. From dictionary.com: 40 (forty) is the natural number following 39 and preceding 41. Despite being related to the word "four" (4), 40 is spelled as "forty", not "fourty". This is because etymologically (and still in accents without the horse-hoarse merger), the words have different vowels, "forty" containing a contraction in the same way that "fifty" contains a contraction of "five". The letters of the word "forty" are in alphabetical order; this is the only number that has this linguistic property in English.
  17. So I picked up a couple of these snom meeting point $700+ phones that look good and sound great, BUT they will not PnP register with PBXnSIP. I can manually register just fine, but the pbx will not respond to any PnP requests from this phone. I also noticed that in the settings area you do not mention the phone at all (e.g. for firmware updates) and I can find no mention of this phone anywhere in the forums. Because I have two of them and they both behave the same way (not registering) should I assume that you guys will eventually release an update which will let me use this phone with PnP, or is something wrong? Should I just do it all manually for now? Or is there some way I can force the PnP to fire? Thanks, Alex
  18. OK, but in other examples you show dial 9 being used in your dial plans - but you have hard coded certain phone models not to allow this behavior? I guess I don't understand why you would do this. So, if I understand your response correctly you're saying I need to manually adjust the dial plan xml file to allow the dial 9 behavior? Two questions: - Where do I put the modified snom_3xx_dialplan_usa4.xml file so it gets pushed to all phones? - How do I ensure this file does not get overwritten by the pbx when a phone registers - as it seems it gets re-generated each time any change is made. - FYI, in the U.S. business market easily 90% of all corporate systems still dial a 9 to get an "outside" line. Any installation of a system like this into a corporate environment is gradual which means I have to replicate the behavior of the current system while I'm cutting over. While it may "simplify" dialing to not use a 9, in many situations it would become a training nightmare as some phones would dial 9 and some would not. Ideally your system should adjust these provisioning files based on the dial plan setup. Currently it's very confusing.
  19. For some reason the dialplan being passed to my snom 300 phones is not allowing me to dial 9 for an outgoing line. When I dial any seven digits starting with anything other than 1, it fails. I looked and the generated dialplan it's pushing to the phone is called snom_3xx_dialplan_usa4.xml Inside, the code reads: <?xml version="1.0" encoding="utf-8" ?> - <dialplan> <template match="1.........." timeout="0" user="phone" /> <template match="[2-7]..." timeout="0" user="phone" /> <template match="8[2-7]..." timeout="0" user="phone" /> <template match="[2-9]11" timeout="0" user="phone" /> </dialplan> Well, here it is. There is no way I could dial 9,1 213 555 1212 - with the above plan. It's set for 1+ dialing. But I have the dialplan set up for 9* and the provisioning is set to USA 4 digits! Is there some place I'm missing to get the pbx to make the adjustments to the above auto-generated file so it would be "91..........."? Thanks for any help. Alex
  20. It would be great if thoughout the the web admin screens you had hyperlinks to the support articles describing what the settings did. For example, in trunks there is a setting called "Send call to extension". I would never have known you could put in variables in this field and only discovered the Inbound calls on trunk? page after looking though the forum for twenty minutes. I had to read it a couple of times before I realized that you were referring to the "Send calls to extension" field. HOWEVER, If you had a link next to every variable poiting to the relevant page. It would save tremendous amounts of time, pain and frustration and I would have been done with this configuration in minutes. Can we poll this? AK
  21. First of all, let me explain that this is a brand new cs410 It is vanilla, out-of-the box. The only thing I did was update it to the latest firmware (Linux) and register a phone. I set up the outdialing rules exactly as specified in your pages, hooked up a standard 1MB POTS line from AT&T to the fx0 port and tried to dial a call - and am getting these errors. I thought either I'm blind or something's wrong, but I could find no mention of any 11 digit feature on any page. I looked at every setting on the PSTN gateway page (reg_ip.htm) Domain Setup (dom_settings.htm) and the trunks page (dom_trunk_edit.htm?trunk=1) nothing says 11 digits. Finally, I looked through all the drop downs and discovered an option under 'Rewrite Global Numbers' within an individual trunk page, called For NANPA (11 digits). Rewrite Global Numbers - huh? Now it works, but I also discovered that now, if I try to put in 9[411|911] in the dialing rules, it causes the cache dialing problem I mentioned in a previous post and keeps dialing the same phone number. But that's for when I have time to deal with it. I'll just make two rules for now. Can I make a suggestion? Can you hyperlink every setting name within the admin pages to the support page describing what it does? It would really help tremendously, be easy to do and would probably really cut down on your support calls. Thanks for your help, AK
  22. I know it's illegal, but 14 year olds with something to prove don't care! Anyway, if an admin is silly enough not to set up email notifications then he deserves what he gets - but hearing that pbxnsip will shutdown immediately when it sees a license violation is a big problem. If you gave us a week, that is plenty of time to fix a problem like that. If I were an admin and got a warning there was a license violation and the system would be shutting down in 168 hours - and one every hour after that - I would be ON THE BALL to get that puppy fixed! Hey, you could also ring the phone of the admin really annoyingly or send him a voicemail too! You know, just creating a log level that sends urgent warnings like that could be helpful for other things too - then all you have to do is flag it if a license violation occurs. Think we can put it in? AK
  23. Fair enough, but this is hanging off an AT&T 5ESS and dial tone is instant, but I did what you requested above and got the same result. Again, my analysis points to your SIP gateway. Log files show that 11 digits are sent and it is stripping off the one when outputting DTMF on the line. I monitored what was sent using a decoder clipped onto the POTS (fx) line. I bet there can be a setting in sipfxo.conf that will control this behavior, but the default is floating back and forth in your units because it is not being explcitly set. Can you please check this? Tests If I send twelve digits by adding an extra 1 at the beginning (112136181000@ the gateway outputs ALL TWELVE DIGITS! If I send 12 digits by adding an extra 0 at the end (121361810000@, we also get all twelve digits, but the call goes through because the phone company ignores the last digit. This gives us a workaround: WORKAROUND I like to set up dial 9 for outgoing calls as it gives more control and most UScustomers are used to it. IN DIAL PLANS TRUNK: PSTN gateway PATTERN: 9* REPLACEMENT: *0 This adds an extra digit (0) at the end of the dial string which tricks the gateway into dialing the leading 1. The phone company will ignore the last digit. When this gets fixed, you won't have to recode your dial plan as nothing will change. The only downside to this workaround is an extra 20 milliseconds of wait time before a call goes out - for that extra digit Not quite... Somehow, using the above method it creates another problem and the same number somehow stays in cache so no matter what number is dialed - from any extension - the gateway always sends the first phone number dialed using this method. What a mess. This had probably never been caught because most people send all long distance calls over SIP trunks and not back over the PSTN. Unfortunately, we have to diall 11 digits for all calls. Help? I have to put this thing in production in less than a week. AK
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