mentor seeking

Discussion in 'General' started by j99991, May 3, 2009.

  1. j99991

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    hello,

    i'm somewhat a new member at this community, though i have been asking questions about elastix. i heard about elastix from another source and come here to get more information about the system and it works. further to my questions i also read elastix without tears which helped me alot knowing more about the system, setup and functions. however while reading the information about elastix and voip in general i noticed that i do have some questions which answers to them i couldn't find at the elastix without tears. i tried to ask at the forum but i now realize that those questions and needs from the system are to complexed and thus need to be solved with direct help and not by forum posts.

    since i believe this is the most experienced place to discuss about the system and since reading many questions and answers of other memebers which looked professional, i would like to ask if there is anyone in this community that thinks he could try and be my mentor or at least to help me with the 3 topics i will need answers for.

    the 3 problems relate to bandwidth coomunications via elastix, elastix functions ( functions that i didn't find answers to at the elastix without tears) and the last regards to elastix systems and special crm needs.

    i hope i could find a ember who will be able to help me solve those problem in order to start using the elastix system.

    thank you
    j99991
     
  2. Mirko87

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    Hi j9991,

    There is no problem... You can ask what you want, and here there will be certanly someone who answer to you...


    I can start to get you 3 links to understand the Bandwidth usage during a VoIP conversation:

    Link:VoIP Bandwidth Consumption
    Link:VoIP Bandwidth Calculator
    Link:Another VoIP Bandwidth Calculator

    For the others... You have to add more precision to your request... and I my opinion you would have to open new topics on a new request, with a "good" title, so that in future another user of the forum will find what He wants looking for the title of the topic...


    Mirko
     
  3. j99991

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    thank you for your reply.

    i understand your suggestion and already did that. i opened few month ago several topics regarding elastix and how to use it. the first topics were easy and i got al information needed. i also opened topics for the harder and more complexed questions i had. one of them was regarind using several ports with elastix, qos and traffic reshapings. unfortunetly, it was hard inserting all information needed regarding those at the messages because there are lkots of information and demands we will need to have. thus, i couldn't find any answers or solution to my problem though the help of some members here.

    one of my most important question is regarding bandwidth, qos with elastix, using several ports for each ext. and traffic reshaping.
    i believe that those problem didn't receive any answer since this is not the plkace to discuss about them because of thier complexity. that is why i was asking and still ask if any one be able to help me with those problems by a direct discussion.
    thanks for your suggestion but i believe that more intens help is needed at this case.

    however, if any one will still insist i will be happy to try and give more information, though i do not think this is the place but... ehy i'm kinda new here so i believe you know better then i do what should be here and what shouldn't.

    thank you
    thank you
    j99991
     
  4. j99991

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    by the way. one of my questions and a more described information regarding the situation can be fun on my post here , port opening seperately for incoming and outgoing,

    thank you
    j9991
     
  5. jaschenck

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    which ports you need to open depends on if you are using SIP or IAX2.

    SIP 5060
    IAX2 4569

    I use both so I have mapped both to my Elastix box
     
  6. j99991

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    ohh i thank you for your reply.

    but unfortunetaly me problem is much bigger then that an reffers to the combination and problems of using the same infrastructure for data internet and voip.
    you can find more information about my problem at my post :port opening seperately for incoming and outgoing



    thank you
    j99991
     
  7. j99991

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    by the way from that post i understood from romanico that the problem could be more with the rtp and not with the sip since i understand that the sip is only for signaling and thus do not use so much bandwidth as the rtp which is used to transfer the voip packets.
     
  8. jaschenck

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    In my case I just setup QOS in my router to give priority to all traffic to my Elastix box so ports are not really an issue QOS wise
     
  9. dicko

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    Many have tried to help you, but until you get a firmer grasp of how IP communications and QOS/Routing work you will still be stumbling in the dark. You keep asking the same questions and getting the same answers! currently with 128k up you are realistically limited to two maybe three concurrent calls with ILBC or g729 given the overhead needed, add two remote offices with VPN and it's overhead and the "tandem" calls you envisage, to be honest "it just won't work" , even with a good QOS in place your data will be at best "turgid". Your problem is no bigger or different than anyone else, so please don't be daunted :) your challenges are the same, but basically if you try and connect a fire-hose to a garden-hose someone is going to get wet! . I suggest that the boys at Palosanto (our sponsors) at

    http://www.elastix.org/index.php?option ... &Itemid=35

    might help you increase your understanding, they offer paid consultation and are very knowledgeable
     
  10. ramoncio

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    In the other thread I gave you some advices.
    If you don't have enough bandwidth you won't be able to use many VoIP simultaneous conversations without huge problems.
    First you need to estimate the MAXIMUM traffic you are going to need for all your wanted remote extensions and offices. If with one internet connection you don't have enough, then you'll need to get 2 or 3 connections and load balance them, at least in the main branch.
    The bigger the spare bandwidth, the less you need to finetune your network, so in your case you will need someone with enough experience and knowledge to install it for you. Ask the Palosanto guys as Dicko tells you. But have in mind that if you dont have enough bandwidth they can't do miracles.
     
  11. j99991

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    thanks for the comments.

    dick and ramonico. maybe i didn't explined this good enough.
    my problem is not with the bandwidth. i know we will need to have a better connectiong and faster bandwidth rates and currently just waiting for better speeds to use.

    again my problem is not the bandwidth. my problem is with the scenario. i'm afarid that too many incoming calls ,which i cant control, cause i can not control how many people call me. i'm afarid that many calls will be answered by the system and by that take too many bandwidth which means i could not use my internet connection for data purposes.

    this can occur even if i had 10mbps or 20 mbps or 50 mbps this is a scenario that is not relevant to the amount of speed you have.

    i know i need to have higher speeds in order to make and receive more calls. but this can scenario can occur to me even if i had those speeds.

    that is why i'm trying to think of a way to prevent it.

    my though was to use 2 prots 1 for incoming calls 1 for outgoing calls and to traffic reshape the incoming calls. by that i can control on the amount of incoming calls, which means i can prevent a scenario in which too many calls disable my ability to use the internet for other things like data or maybe even trying to make outgoing calls.

    i hope you understand my problem now.

    what i didn't know and was trying to ask here is first if elastix can support 2 ports. to define all extention to use 1 port for incoming calls and 1 for outgoing while making this port to have a maximum bandwidth usage. since i do not know the system i had to ask it here. if the system can do or support it and if you think it's possible.

    again i hope you can understand my question regaring this.
    i'm only trying to prevent this kind of scenario or situation.

    dicko, thanksd for the link i will go to it right now.

    but pls i will be happy if you can try and tell if this is possible with elastix.

    thanks
    j99991
     
  12. j99991

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    by the way romanico i thank you for explining my what is the rtp i understand that this is the most important issue and not the sip since the rtp is what's acctually loads the voice.

    so i understand i need to also cosider also the rtp in order to have a good solution and that what's acctually should be reshaped is the rtp ports.
    however i do not know how the calls are being done.

    for example,
    can i tell elastix to use port 5060 on all extentions to make calls and to use port 5070 to receive calls. and then just to define maximum bandwidth usage for port 5070 so i could receive only x amount of bandwidth which means y amount of calls?


    thanks
     
  13. j99991

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  14. dicko

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    NO, it just doesn't work that way!!. As I say until you invest a little time in learning how IP communications work, in this case SIP voip, you will remain unenlightened.
    One More Time, and with feeling!!, 5060 is the port normally used for signaling (i.e. maintaining presence and accepting connections), the vast majority of traffic (RTP, the audio) is on one port per audio stream usually between 10000 and 20000 arbitrarily chosen when each call is set up. if you get too many calls, just don't answer them, no audio= almost no bandwidth. RTP stands for "Real Time Protocol" it is probably the only protocol you really Do NOT want to, in your words, "reshape"

    I refer you to
    http://www.faqs.org/rfcs/rfc1180.html
    and
    http://www.geocities.com/intro_to_multimedia/SIP/
    and ultimately of course
    http://www.google.com

    Please, PLEASE spend a little time reading them.
     
  15. jaschenck

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    This seems to be getting rather pointless, but I got some extra time on my hands so what the hell I will give it another go.

    You say you have no idea how many incoming calls you will get.
    well I guess none of us do when we get a new DID (phone Number) but I do know how many people I can talk to at one time.
    Why don't you worry about the over flow of incoming calls when it happens? You can limit the calls per trunks if need be or as Dicko said just don't answer at some point. how many phones are you planning to buy anyway? and how many lines per phone?

    I think what you need to do is stop posting for a bit and start doing, then post with you problems after you try and work them out for yourself.

    That's what I did about 9 years ago I did not have a clue, but I found some retired Dlink MGCP ATA's on Ebay down loaded Asterisk and installed it on a Trustix box I had. It took me over a week just to get a dial tone but I worked it out.

    That's how you learn jump in with both feet see what works and what does not. Some people get it in the end and some quit but if you don't step up to the plate you will never hit the ball that's for sure.

    I still don't know it all I have Two Cisco 7961s that give me headaches and cause me loss of sleep, but that's just part of it. If you want simple call AT&T and order some lines and a phone system, or pay some one to build something for you. If you want to go the route of the roll your own VOIP then my friend it's time to stop posting and start rolling because until you do you will never get it.

    You have come to the point here where people are just going to tell you RTFM and to be honest I am truly surprised that no one has gotten rude with you yet.

    So here is my TODO list for you.

    1. build your Elastix box
    2. get your phones
    3. find yourself a provider

    Try and get it to work while reading and then if you have issues ask good questions and I am sure you will get quality answers.

    Best of luck to you

    Jim
     
  16. dicko

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    Amen Jim,
    One can lead a horse to water, but if it only sees grass it still has to learn to swim to get to the other side. (mixed metaphor to the second derivative on purpose!)
     
  17. j99991

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    well i see.

    dicko and him thanks.

    i'm sorry i was repeatedly going on and on with my questions it just that it was hard for me explining exactly what is it that i want to do or to avoid of.

    dicko, i understand that as long as i do not answer or pick up the phone there will be no use of the bandwidth since there is no audio.

    i would just like to ask what will happen in case i dewfine an ivr?
    does ivr count as a voice as if i answered the phone?
    i do mean to use an ivr and that where it all came from . i could have just not answered the phone but since the ivr pciks up the call automatically i just didn't know how to over this kind of a situation.

    by the way dicko i'm logging right now to the sites you gave me in order to read more about connections and sip so thanks for the links.

    jim, you mentioned that at each extention i can define a maximum number of calls. i beleive that could be the answer to my problem i just didn't know this is possible. i would just like to ask if i could also define it in the ivr as well ? or only at regular extentions?


    again i'm very very sorry for botherring you with these question i hope that now i could find a solution for this. and jim, i do belive that you are right i would just need to install it and to see what will happen. otherwise i wont know.

    i hope you could answer those question while i'm searching for more inforamtion regarding connections.

    thanks again
    j99991
     
  18. ramoncio

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    You can define a maximum number of simultaneous calls per trunk. This should be enough.
     
  19. j99991

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    romanico thanks for your help i beleive that this is the solution for my problem and this is what i wanted to know.

    i understand that this could be done also on the ivr extension.

    i'm writing this just after finishing reading the links dicko attachmed me here so thanks dicko. i also read about rtp and not just the sip and i believe it helped me alot.

    would just need to know if this solution is also applicable to ivr as well since this will be the one to answer the calls.

    thanks
    j99991
     
  20. jaschenck

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    I clearly stated that you can define max calls per trunk.
    being that you are asking what about the IVR tells me that you have not read much or just don't have a clue and maybe this is just too much for you.

    Call flow should you have an IVR would be as follows

    provider-------trunk-------IVR----------extentions

    it could be more complex then that but that should give you some idea
     

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