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Alex Kasperavicius

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Everything posted by Alex Kasperavicius

  1. I have exactly the same problem. CS410 ( (Linux)) and whenever making a call over the PSTN , the box will not dial the one. I have put a butt set (test set) on the line and confirmed that the system is sending only the area code and number. We are in California where every call must be 11 digits. Here is log: [5] 2009/07/15 18:34:00: Dialplan Default Dialplan: Match 912136181000@sip.scrumption.com to <sip:12136181000@;user=phone> on trunk PSTN Gateway Note that the PBX is sending eleven digits to the internal SIP adapter and that's where it seems to be stripped. I think this is unique to the CS410's internal adapter.
  2. Would it be possible for you to write up the best practices method for provisioning snom phones using pbxnsip? I have a box of brand new snom phones and a cs410 and am getting a little frustrated with the amount of time its taking to run simple software upgrades. snom is advising logging into each phone and pasting a URL into the phone. This upgrades but it's silly and very time consuming to have to do each one by hand - and it sticks at the end and requires a manual re-boot of the phone! I would *love it* if you had an article which told me: - WHERE on the PBX to drop the snom software and default settings file; and - WHAT settings I need to change on the PBX So that all I have to do is know the MAC address of a new phone. (It's printed on the box and on a sticker. ) I put that in a PBXnSIP account record, plug in the phone, it boots, upgrades to the latest firmware, loads its settings and I'm done. A cheat sheet with the basic steps to make this happen would be gold - especially if you continually updated it as the steps change. We would always know where to look. Can this be done or am I blind and you will point me to a page I have missed? Thanks, Alex
  3. On a related note, how come the default behavior is to brutally shutdown the entire system when a license goes over? Wouldn't it make sense to allow only X extensions to work with a log message? Or better yet, a warning message emailed to the administrator with some chance to fix it? Why so harsh? Consider: if a hacker were to crack PBXnSIP, they would get rid of the license module completely - this only inconveniences your paying customers. AK
  4. Agreed - this feature needs documentation as it's a big black box mystery. Mine turned off and the only way to get it back was to re-install the software. Ahh! AK
  5. Can't you guys just regenerate the license? This seems like it shouldn't be a problem needing to be solved here. I am sort of perplexed at this response from pbxnsip and would be more than perplexed if I were in the poster's position. License problems which incorrectly limit functionality are always on the developers to solve - like immediately - right? They have a tendency to blow up - and quickly. Just hearing this puts a knot in my stomach and makes me re-think purchasing any of the things because I don't want to be in the same "gee whiz" position wasting my time trying to get around an incorrect license block. It's a big deal for me anyway. Also, I guess I just assumed a CS410 would support 4 trunks and 10 stations out of the box. If this is not the case, I am perplexed as to why this particular model number was chosen. AK
  6. Automatic Gain Control works pretty well, but in a conference there's always the person who has it on speakerphone and is typing - loudly, has someone come up to them and start talking without realizing they are on the phone, or is at home with a dog barking or is doing the dishes. Most conferences are with sales people - and while I agree that Joe Average Sales guy is never going to learn features, muting is pretty key. In many systems, you hear "to control your mute, press *6 at any time" or "Press *1 at any time for a list of commands you can use" right before joining. As an admin on the call, you need to be able to mute everyone when no one will fess up to being the noise maker! then anyone who needs to speak just presses *6 to unmute themself. Conferences can get pretty hairy sometimes. An xml based phone thing would be cool, but if you're already creating the xml why not also create a web page on the fly for the administrator? Webex and others do this - and you can see a list of all participants, right click anyone to mute them or dump them AND you can see each person talking: when they speak their icon makes little sound waves. VERY handy! When setting up an ad hoc conference, you already have the email of the user - you could send them a link to the page, make it pick-able from a list of current conferences (with PIN for admin rights) at http://<server>/conference and even make it predictable (like http://<server>/conference/<conference number> so it could be put in emails easily, etc. This is what the WebEx iPhone controller looks like. In it you can see that Grace is the moderator, calling from a phone only and is muted, Jan is on a computer and currently has presentation rights, Joe is on a computer and can speak, Zoltanis on a phone only and can speak. Interesting what can be presented in a small space, eh? AK
  7. I think the trick with Wikipedia is to go in guns blazing, put in a lot of information with a lot of references, don't make it sound too commercial and then monitor for a few weeks. There's always someone with an axe to grind, but if you follow all the rules there's not much they can do. pbxnsip is a legitimate system, so it should stay up. I think what may have happened is others may have been spooked. In 2006 a page called pbxnsip was deleted - so attempting to recreate it and/or looking up pbxnsip showed that it was removed for cause and set off all sorts of dire warnings if you attempted to re-create. But - looking at the old logs it was only because the page was empty. Anyway, it's up now and all should be good. By the way, it would be really nice if you guys opened your wiki. I know it feels weird, but keeping it locked down is preventing anyone from making little changes that could be helpful. The wiki model really does work for stuff like this, but it has to open or you lose the benefits. AK
  8. Hi All -- In a fit of something or other I made a Wikipedia page for pbxnsip. Please check for errors/additions! Here is the link pbxnsip on Wikipedia AK
  9. It's some home brew system that a vendor cooked up and lives at their site. We just use it. These features seem like they would be pretty trivial to add and would really add a lot of value to the system. PBXnSIP guys? Any thoughts?
  10. Hi All, Here are some controls from other systems that you might consider if you're feeling inspired! : User Features Press *1 to hear a help menu. Press *0 to reach an operator. Press *6 to mute or "un-mute" line. Press *4 to increase conference volume. Press *7 to decrease conference volume. Press *5 to increase your voice volume. Press *8 to decrease your voice volume. Moderator Features Press *91 to hear a participant count. Press *92 to hear a roll call of participants. Press *93 to disconnect all participant lines. Press *94 to lock or unlock conference. Press *95 to dial out to participants. Press *96 to mute all participant lines. Press *97 to un-mute all participant lines. Press *21 to activate Subconferencing. Press *22 to initiate record and playback (*22 again to pause/stop the recording) Press *31 to turn Conference Security Code on/off. Press *32 to record your Conference Introduction. Email at conference end of all participants with caller ID, call in time, hang up time & length of stay. Any thoughts, comments? I would love to see these as I could dump the system we currently have and the cost savings on conferencing alone would pay for PBXnSIP!
  11. So gave up and wiped & re-installed - now it works fine. PBXnSIP sees the request come in from the phone and dynamically generates the correct CFG file on the fly; Registering any extension containing the phone's MAC address in its record and adding them as lines on the phone in numerical order. It seems that what I first did, putting my own configuration files in the TFTP directory, permanently broke the Plug and Play functionality and I had no way to restart it. After deleting the files from the TFTP directory I saw the requests coming in on the log file, but the PBX's response was that the requesting MAC was already being used in another domain (yet there was only one). This really needs to be detailed in the Wiki as the ONLY method it mentions is copying the example config file on the wiki and dropping it in the TFTP folder. It doesn't mention plug and play *at all* -- If I have some time in the next few days Pradeep, I will send you a wiki markup file. Alex
  12. Alright, I have a few more answers. In the log file I am seeing this: [3] 2009/04/22 16:11:12: MAC 001562EA58FE already used in different domain <--- That's good news - BUT I only have one domain. So I have deleted all domains on this test system and recreated one from scratch. I also deleted old configuration files from the folder. - but no go. How do I trigger PBXnSIP to create configuration files? AK
  13. That's good to hear, but I still don't understand how this works and the wiki page is pretty sparse: (http://wiki.pbxnsip.com/index.php/Cisco) indicates that "You can register more lines by properly filling in the lineX_<parameters>" in the CFG file for the phone. I guess maybe I'm assuming that you have a feature that will automatically add lines to the phone and/or dynamically modify the CFG file. Currently I am manually editing the CFG file for each phone - which seems silly. If I don't put a CFG file in the TFTP directory and just leave the CISCO files there, the phone remains unprovisioned. Am I missing anything? Thanks much, A
  14. Hi All - I have a couple Cisco 7960 phones and would like to use the PnP functions to program them. In the wiki it seems to indicate that this is not possible and one must manually change the CFG files in the TFTP directory, which is what I', doing now. But in some other posts it seems like PnP is possible with a 7960. In any case, I can't really get my head around how the PnP function works and wonder if anyone can describe the basics or point me to an overview. I can't find anything. For example, I see the PBXnSIP is making configuration files, but they are not in the TFTP directory and I have no idea what would either trigger a move or redirect the phone to somehow get them. Also, they are xml and I don't think these 7960's support them. In any case, if someone has made their Cisco 7960 phones auto configure using the PnP feature, would you mind posting how? Thanks much, Alex
  15. Hi All - I am looking at rolling PBXnSIP at remote location which already has a few POTS lines in place. While I had considered using the CS410, I would prefer to use our own server equipment as our team is trained on them, they have backup processors, power, etc - (...and they are already installed and in place!) Has anyone had good luck with a specific brand of VOIP gateway or is one kind recommended for use with PBXnSIP? I found a long list at: http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways ...but it's a long list and I have no frame of reference. Again, I only have 5 or 6 POTS lines and just want to make them usable by PBXnSIP. I imagine this box would simply make these POTS trunks available as SIP trunks. Suggestions anyone? Thanks, Alex
  16. Had the same problem and setting Remote Party/Privacy Indication to RFC3325 (P-Asserted-identity) fixed the problem. Note this is not mentioned in the Wiki instructions for setup of Exchange interoperability. Might you guys consider opening up your wiki to registered users? I mean, some of it is pretty sparse and I for one would be happy to add little corrections here and there. It certainly would be easier for you guys to monitor additions than having to type all that stuff yourself. I think you would be pleasantly surprised. AK
  17. I have tried every possible combination I can think of and in every case the P-Asserted Identity field sent by PBXnSIP is the same thing. It never changes. I really think this is a bug. I should be able to set the identity to my "from" phone number but it's always the username for my SIP account. Should I post clips from the sessions to show what I am describing? AK
  18. If someone is spoofing caller ID for nefarious purposes, the fault (and legal trouble) lies with the person doing it, not the PBX provider. Your decision to limit functionality to avoid this scenario actually puts you in more legal danger: Someone could figure out another way to use your system for illicit gain and you could be sued for negligence because you didn't limit that! I have seen and used this forwarding feature on other systems. It's pretty standard and very helpful. Would love to see it. Also, spoofing CLID can be done in so many ways (see spoofcard.com) if someone wanted to do it. AK
  19. Hi Guys -- Have been playing with your software for a couple of days now and am quite impressed! Very nice job! I am running into a bit of an issue with outbound caller ID to a PSTN SIP trunk provider (flowroute.com). Using Wireshark I discovered that with every outbound call, the P-Asserted Identity is <SIP Username>@<domain> instead of <Account field>@<domain> or even, ideally, ANI@<domain>. My understanding is, and I confirmed with flowroute, is that the P-Asserted identity field should be the 10 or 11 digit ANI (Caller ID)@<domain> unless the privacy bit is set. It is instead being set to my flowroute account number which is problematic. I have been playing with settings for hours and even spoke to the guys at flowroute and they insist that the P-Asserted Identity field is supposed to be the phone number. Is this a bug, are they on crack, or am I missing something? I have been trying to make this work for hours and could really use your help. Thanks much, Alex
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