Left high & dry by Asterisk tech

Discussion in 'General' started by sskiba, Aug 4, 2008.

  1. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    I'm not sure this is where I should post this or if anyone can help.

    We were sold a linux box with Asterisk & all that we needed to have the system working. The system is working in a very minimal configuration. The tech led us to believe we would be able to install a GUI & we would be able to make needed changes to the phone system ourselves. This was over 4 months ago. We've since discovered that is not the case. Our understanding is that we would need to install a different version of linux etc to handle the GUI. The phone system is critical to operations and we don't have the linux expertise in the office to do this and be able to guarantee that the system will be up and running the next day. We've been looking at elastix for the gui & even have an additional box configured to work with elastix. Our concern is that if we try the upgrade and it doesn't work we would want to be able to go back to the minimal working configuration.

    Any thoughts, ideas or suggestions would be greatly appreciated.
    Thanks, Steve
     
  2. packetfish

    Joined:
    Jul 3, 2008
    Messages:
    10
    Likes Received:
    0
    Howdy Steve...

    First, relax, chances are you have a base distro of Linux and Asterisk without a web front end installed. You'll need the required credentials to the box (you need the root password to the server), you should be able to easily add a web-gui providing you have it.

    If you don't and your consultant didn't provide and won't, you may need a lawyer... or you may wish to pay someone just re-build your system from scratch designed around what you need. Since you didn't specify per se you were running Elastix or something else (I suspect something else because Elastix usually installs with a web front end), your mileage may vary.

    Anyways...here's possibly some info to get you going...



    In Fedora/Redhat/CentOS releases you'll probably want to...

    - login as root
    - yum install freepbx (as root)

    In Debian based distros (Ubuntu, Debian, Freespire, etc)
    - login as root
    - apt-get install freepbx, or aptitude install freepbx (as root)

    FreePBX is the web front end you were hoping for and didn't get from your consultant/jerkazoid that didn't bother to install it, probably as a lame attempt to lock you in to support via him only.

    You can read more about Asterisk or FreePBX at their respective sites. I hope this helps and good luck to you!
     
  3. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Hey packetfish,

    Thanks for the reply. I think you're exactly right. I think that our consultant/jerkazoid (good description) was trying to keep us locked to his support because there was no interface provided once things were set up. Anyway...I really appreciate your help. I've been looking through the FreePBX site & learning a lot.

    I think we're going to be in good shape. I checked our linux version & it shows - Linux version 2.6.9-67.0.4.EL (mockbuild@builder6.centos.org) (gcc version 3.4.6 20060404 (Red Hat 3.4.6-9)) #1 Sun Feb 3 06:53:29 EST 2008

    Also checked CentOS & it shows CentOS release 4.5 (Final)

    Early next week we're going to give it a try. I'm thinking we'll try & walk through the installation procedure instructions for centos 4.3 on the FreePBX site.

    Does anybody see any red flags with the information that I'm showing at this point?

    Thanks, Steve
     
  4. MageMinds

    Joined:
    Jun 26, 2008
    Messages:
    55
    Likes Received:
    0
    Installing FreePBX over a working scripted Asterisk might break everything, so be prepared to reconfigure everything yourself, Trunk, Dialplans, Outbound and Inbound routes, Extensions, etc...

    If you're in Elastix forum, you might consider downloading the ISO and install from scratch a system build with GUI in mind if things go wrong... So try to install FreePBX but if you see that nothing is working, try to get as much information as possible about the current config. Reinstall from scratch Elastix and reconfigure... Elastix integrate not only FreePBX, but other subsystem that make it interesting to small and medium business, like a Fax server, a IM server, CRM Server, etc...

    DURING THIS TIME YOUR PHONE WILL NOT BE WORKING UNTIL YOU RECONFIGURE
    DURING THIS TIME YOUR PHONE WILL NOT BE WORKING UNTIL YOU RECONFIGURE
    DURING THIS TIME YOUR PHONE WILL NOT BE WORKING UNTIL YOU RECONFIGURE
    DURING THIS TIME YOUR PHONE WILL NOT BE WORKING UNTIL YOU RECONFIGURE

    Have I said it enough time?

    DURING THIS TIME YOUR PHONE WILL NOT BE WORKING UNTIL YOU RECONFIGURE

    I think so!

    Talk us about your present configuration, ATA Adapters or VoIP Phones? FXO Cards or VoIP Provider? More than one provider? IVR? Everything your system is doing, we can help you locate the config file, if you can access the server of course.
     
  5. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Wow! Didn't know I'd get logged off automatically if I took too long to write my reply. I had it ready to go, hit Submit & lost all of it.

    K so it kind of sounds like I'm gonna be screwed here. I guess I won't just willy nilly download & install FreePBX on this thing. So I'm leaning heavily toward doing a complete reinstall from scratch using Elastix & then reconfigure. I'd like to get to know this system from the bottom up.

    Here's what I know about the system so far:

    Intel MB D201GLY2
    512M RAM DDR2
    4Gig Industrial Flash
    2 Polycom IP 330 phones
    3 Polycom 601 IP phones
    Lynksys 24 Port POE and QOS
    Linux version - Linksys 24 Port POE and QOS
    Red Hat 3.4.6-9
    Asterisk - 1.2.26
    FXO Card - Don't know & don't know how to check for it.
    VoIP Provider - CBeyond
    We do want IVR capabilities
    It is a dedicated box & we have a dedicated T1 for the business.

    I am able to access the server although I suck at the command prompt. I can use a cheat sheet to get around some.

    I'm guessing since we're using the Polycom IP phones we're not using ATA Adapters.

    Do you think it's feasible for an inexperienced Linux user to try & do a rebuild & configure the system in an evening? Probably a really long evening! I was told by another Asterisk user that I'll likely want to backup my etc folder at a minimum. I've attempted creating copies of all the folders on the system & was only partially successful.

    I'm really interested in hearing more to see if I can get this accomplished. We've had the system for over 4 months & no one has been able to get it beyond minimal functionality. I appreciate all your help & look forward to hearing more. Thanks, Steve
     
  6. packetfish

    Joined:
    Jul 3, 2008
    Messages:
    10
    Likes Received:
    0
    I think you will be just fine. Although its probably more work than you wanted, you probably will be better off with a fresh install of Elastix so put on that sys admin hat and get ready to learn a little.

    I use basically the same setup as you have: Polycom 320 phones, I use Dell and Zxel PoE switches.

    Grab 2 things to make this easier on yourself...

    - Elastix without Tears guide (you'll find links on the Elastix site or Google for it)
    - voip-info.org -- you'll find a good guide for setting up the polycom phones there

    Additional hints/tips:

    Polycom phone firmware is available from the trixbox (another pbx distro) yum repo. Polycom does not provide firmware without a customer contract/agreement so trixbox repo is a good place to go (unless Elastix now is including, which last time I looked they weren't) to get firmware for your phones.

    Cbeyond should be able to help you get your SIP trunk configured once you have Elastix up, in which case you can immediately test with a softphone like x-lite, then move on to getting real phones configured....

    Good luck with recovering and getting a system up. Come back to the forums and I am sure many would be glad to help you get through this if you have additional problems.
     
  7. MageMinds

    Joined:
    Jun 26, 2008
    Messages:
    55
    Likes Received:
    0
    As packetfish said grab the Elastix without tears, follow the step ... when you search on the Internet to find help you have to search for solution not for Asterisk, but more specifically for FreePBX, sometime it's easier to understand, since the examples you will find will use the same terminology as FreePBX ...

    For the T1 is it a real T1/PRI a fiber optic or it's a VoIP PRI, so your voip provider provide you with 23 voice channels?

    If you have a real T1/PRI you will have to configure that, I can't help you I have no experience with that and Elastix, mabye someone can drop a line about how easy it is to configure that card.

    FXO card are if you have a hard PSTN line connected to the server, but I doubt it if you have a T1/PRI.

    Have fun and if you don't know some basic Linux find someone that does, that might come handy if you have trouble with your T1 controller card.

    You will have an easy access to IVR after Elastix is installed, in fact I have Elastix at Home and I have an IVR for when I'm not there, callers have the choice to leave a message or call the cell phone, then if the cell phone doesn't answer they end up in the Elastix voice mail. I have hidden options for callback to DISA and direct DISA feature into my IVR also ... I love my Elastix I can do whatever I want with it.

    Good luck and where here to help if you have touble!
     
  8. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Hey Guys,

    Thanks for your replies. I'm gonna go ahead & go forward & see what happens. I'm going to attempt putting together a Linux box this weekend with Elastix just to get a feel for that aspect. I think the only way we're going to be able to do this is just start from scratch one of these evenings when the phones are quiet & work on it until it's good since we don't have additional equipment to play with.

    Thanks again & I'll probably be back on soon with lots of questions if we can't get things going pretty quickly.

    Steve
     
  9. Ichorcom007

    Joined:
    Apr 16, 2008
    Messages:
    52
    Likes Received:
    0
    Hello Sir,

    My name is Kevin , I own Ichor Communications Inc. out of Toronto. I am sad to hear about your tech. I can have your phones up and running in a few hours with a new install. I have been installing asterisk systems for 5 years and also been installing Elastix systems for 1 year. I currently have about 50 businesses and call centers running Elastix.

    Please check my website for pricing www.ichorcom.com I can have your servers up running freely the way you want in less than a few hours. you can email me at kevin@ichorcom.com. Leave me your number in the email and we can talk. Also you may check out my website at www.ichorcom.com. Since you came through the community I will give you a discount on your install compared to what is on my website.

    Let me know

    Kevin
     
  10. Ichorcom007

    Joined:
    Apr 16, 2008
    Messages:
    52
    Likes Received:
    0
    I can also provide you back up while I work so your services do not go down.
     
  11. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Hey All, I've got a couple of new questions on our Asterisk upgrade. We have a second box now & have done some testing but I need to know some details on what I'll need to try & use this box to switch out our production box. If this one doesn't work we can replug in the working box. Right now the test box has Elastix running & we're able to test it with a softphone. What SIP card would you suggest for a small business that would be running between 10 to 20 consecutive calls? I've looked a little at the Digium 400P & 800P. Do I need to go the Digium route or are there some other options? We will be using only IP phones so don't need analog capabilities. Also what kind of capabilities will we need in a PCI card or will we need one at all? I'm clearly still a newbie. Basically what I'm asking is what other pieces of hardware/software will we need to make this test box work as a temporary production box until we can reconfigure the present working box with the latest version of Asterisk & Elastix & get it back up & running?
    Appreciate your help again, Steve
     
  12. Lou1z

    Joined:
    Aug 20, 2008
    Messages:
    57
    Likes Received:
    0
    for your information, you do not have to worry about passwords with linux. simply boot into runlevel 1 and change the password via the cli.
    that will get you root access to the box. you may even find that you can boot into runlevel 5 and the grub has been edited to go to 3.
    now you could install freepx over it asterisk but i would say why bother. your installation is minimal and elastix is more than capable of handling it. it will also give you a nice easy learning curve into it.
    i think you have done right by putting it onto a seperate box and testing. now all you need to do is have all your voip details, network setup etc all to hand to do the lot. probably over a weekend?
    as for cards, what are you running? pure voip? no need for one. pri's? there's digium (never had a problem with them), openvox (some call them digium knockoffs but there are good reports on them), sangoma's & rhino's are probably rated as the best.
     
  13. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Moved forward somewhat. We got the test box built & configured. We've got the elastix's gui now and have done a bunch of configuring & reconfiguring but can't get any inbound or outbound calls to work. We were able to look at the working box "sip.conf" file to double check settings but still no success. We set up the extensions & a SIP Trunk with what we believe is the correct configuration.

    PEER Details:
    username=7196323100
    type=peer
    secret=************
    outboundproxy=sipconnect.den0.cbeyond.net
    insecure=very
    host=sipconnect.den0.cbeyond.net
    fromdomain=sipconnect.den0.cbeyond.net
    ;dtmfmode=inband
    dtmfmode=auto
    context=from-itsp
    caninvite=yes

    Nothing in USER Details

    The Register String is exactly the same as the old working box.

    Any thoughts or ideas to help us get past this bit? Appreciate any help. Thanks, Steve
     
  14. MageMinds

    Joined:
    Jun 26, 2008
    Messages:
    55
    Likes Received:
    0
    Check if the register is working ...

    Goto the Asterisk CLI

    # asterisk -vvvvvvr

    CLI> sip show registry

    If it is registered then, try a call and check the screen you could find some clue why it's not working.
     
  15. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Hi MageMinds,

    That helped a lot. So the new box is not showing as registered. What do I need to do to get it to register with our host (cbeyond)?

    Thanks, Steve
     
  16. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    One step forward but still not there!

    So we got some good help from our host & he was able to get us so that the new box shows as registered but when they would prompt an invitation they came up with a 404 error. His comments were that something within our PBX didn't know what to do with the number. We set up a "Direct DID" number within our extension configuration but still no luck.

    Any other ideas as to why we might be getting this 404 error from the host?

    Thanks, Steve
     
  17. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Another step forward!

    We got the 404 error taken care of by resetting all the extensions. Somehow they were screwed up. I also managed to get outbound calls going this evening. I think it was because we had the dial plans configured incorrectly. I'm having no luck getting inbound calls to work however. I've tried numerous inbound route configurations to no avail. Any thoughts from anyone.

    Thanks for all the help so far. Steve
     
  18. MageMinds

    Joined:
    Jun 26, 2008
    Messages:
    55
    Likes Received:
    0
    You'll have to say more about your VoIP configurations, do you have many DID, the DID you receive when you get a inbound call is the string you have put after the "/" in the register string.

    login:password@sip.server.net/did

    With a single register it's possible to receive many DID, but you'll have to hack into the FreePBX configurations files yourself to extract the "To" field in the SIP INVITE packet. I don't know exactly how to do that, but I came across that configuration some times ago, I don't remember if it was on the forum or elsewhere.

    The first time to test your setup you create a inbound routes with "any DID / any CID" and set the destination to an extension or better a ring group.

    After that when you can receive inbound calls, you create a inbound rules with the DID you put at the end of your register string... and you remove the "any DID / any CID" rule. If you need to receive more than one DID from the same register to your VoIP Provider than search about that, there is something to do in the customs config files of FreePBS as I said...

    MageMinds
     
  19. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Hey MageMinds,

    You lost me on a lot of what you said in your last post but we ended up using elastix paid support & they got us through the issue. They spent a lot of time trying different configurations to get the inbound calls to work. It had to do with the settings our host required & wasn't easy to find. So now we have outbound & inbound calls working.

    We seem to have a conflict however with our extension numbering. When we set ring groups we get unexpected results with the extensions. We have a ring group set to ring ext 220, 223 & then 224. We end up with them ringing other phones and in a different order. It's as if the extensions have been re numbered somewhere else & those settings override the extension numbers we see.

    Has anyone had a similar situation? We haven't found anything that tells us the extensions are set anywhere else. Sure would appreciate any help. Thanks, Steve
     
  20. sskiba

    Joined:
    Aug 4, 2008
    Messages:
    45
    Likes Received:
    0
    Success finally!

    We have the new system up & running (still working through some minor tweaks). We took care of the extension conflict by resetting all the phones to default settings & then adding our ip. Then we just reset everything in endpoint configuration using the ip's the system provided. Pretty slick.

    Thanks for all the help on getting us through this. It's a good forum. Steve
     

Share This Page