sip proxy with elastix?

Discussion in 'General' started by souxinh, Apr 6, 2010.

  1. souxinh

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    Hi all,
    I'm new in elastix (about 3 weeks). I setup elastix and calling card 3 weeks ago and everything work great! Thank you elastix people that make it easy to install and setup.

    *Edit*
    My question was that, I have elastix setup with few extensions. When I setup x-lite for my extension, it ask for sip ip/server: I enter my elastix ip address (ie. 192.168.x.x) and it work internally. I just wonder that Is it possible to setup x-lite from other network other than within my lan network? I saw that voip provider have sip proxy registrar like: sip.voipstun.com, or sip.something.com. Can I setup something like that with my elastix box?


    Anyone have any suggestion?

    Thank you very much
     
  2. Siu

    Siu

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    Re:mobile sip phone

    I hope it can help you, please see attachment.
     
  3. souxinh

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    Re:mobile sip phone

    Sorry, I can't see attachment.
     
  4. souxinh

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    Re:mobile sip phone

    Sorry again, I saw it in my email...Thanks
     
  5. souxinh

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    Re:mobile sip phone

    nope. I do not see the attachment. Please reply again. Thank you
     
  6. Siu

    Siu

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  7. souxinh

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    Re:mobile sip phone

    Thanks Siu,

    My first post was not so clear. For example, the pdf that you attached when setup, it ask for voip provider proxy address. I want to know If I can setup my elastix to act like one? Can be done?

    Thanks
     
  8. dicko

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    Re:mobile sip phone

    First off welcome here souxinh:

    Asterisk Proxies for itself quite adequately STUN is normally not necessary in a properly configured system, just read "Elastix Without Tears", set up your sip_nat.conf as explained in the aforementioned document, and register your soft-phone against the external IP address or dns name as appropriate of your server.

    (then read the other posts around here to get x-lite actually working for more than 30 seconds :) )

    dicko
     
  9. souxinh

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    Re:mobile sip phone

    Thank you dicko.

    I have read "Elastix Without Tears" before I set everything up. I will try to explain my systems.:blush:

    First, I am using the dyndns.com service for my external ip.

    I have systemA, runing IIS7 with squid ( on the same system to serve multiple web sites that I hosted ) (ie. a.dyndns.com -> squid will route to web site a:81,
    b.dyndns.com -> squid will route to web site b:82, etc..)

    Now, SystemB - Elastix, and I use g.dyndns.com for my external ip. Squid route this traffic to my elastix internal IP admin page.

    All required ports are open for SystemB. and set sip_nat.conf as follow;
    nat=yes
    externip=nnn.nnn.nnn.nnn
    externhost=g.dyndns.com
    localnet=192.168.3.0/255.255.255.0
    externrefresh=120

    SystemB, already setup incoming DID and outbound routes, and 3 extensions for my laptops within lan network (using x-lite 3.0). Everything work fine, no issues:)

    Also, SystemB already setup for calling card with only one testing customer (me) and I can call into my DID which then, the a2billing asking for pin, authenticate, and route the call out to appropriate dialing plan trunk and to appropriate destination. (All these by Elastix teams!)

    Now, I try setup x-lite 3.0 at my work....it can't registered with my elastix at all.
    No matter what I use (ie. g.dyndns.com or my router WAN ip: 207.xxx.xxx.xxx or my local ip address (desperately :blink: ))

    If I can get this working, I can use mobile softphone on my cell (I have full data plan ...unlimited internet access...) to register it with my elastix and make call to anywhere in the world....(what a plan, ehh???) May be this is just a dream or can it be done?

    Any suggestion?

    Thank you
     
  10. dicko

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    Re:mobile sip phone

    SIP and the ensuing RTP (audio) must traverse all routers "elegantly" , I would check your "work" router.

    sip debug ip <ip to check>

    from the Elastix box will help, if you don't see sip conversations between the two boxes and the IP is your "work" routers external address, then you need to fix that, if you do and the registration is failing, then equally, you will have to fix that also.

    dicko

    p.s.

    I note

    externip=nnn.nnn.nnn.nnn
    externhost=g.dyndns.com

    you can't do both, if your externIP is not static than use externhost, if it is then use externip, if externip is not "publically available", it just won't work :)
     
  11. Siu

    Siu

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    Re:mobile sip phone

    Souxinh sip_nat.conf not affect client register, it only affect internal client talk to external client. do you use one External IP for two dynamic domain?
     
  12. dicko

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    Re:mobile sip phone

    Siu:

    I believe your advice is incorrect, sip_nat.conf absolutely affects External client registrations, and totally controls A ) whether an external client can register and know that it was successful in that registration, and B ) whether the audio will eventually be able to pass bidirectionally after a successful SIP INVITE, assuming your NAT/Firewall boxes are all correctly set up.

    Do you disagree?

    dicko
     
  13. souxinh

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    Re:mobile sip phone

    Hello,

    Thanks for quick respond.

    Siu: yes I use only one dyndns.com (external ip) for all other internal ips (web server, elastix server, mysql server, and soon to have media server)

    Dicko: Sorry, I double checked and yes, I only have externhost=g.dyndns.com not externip.
    for "if you don't see sip conversations between the two boxes and the IP is your "work" routers external address, then you need to fix that" Do I check during my extensions conversation or during did to outbound call conversation? Please let me know.

    and "sip debug ip <ip to check>" this would be my externIP? correct?

    Thanks

    Souxinh
     
  14. dicko

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    Re:mobile sip phone

    Start with

    help sip set debug

    You can check as often and as long as necessary, you can also check all and any appropriate <IP's>, it's your box, only you know what to check

    dicko
     
  15. Siu

    Siu

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    Re:mobile sip phone

    Dicko
    Sip_nat.conf is blank, external client can register and dial-out, but can not hear any voice from other client.

    Souxinh
    Two Elastixs Boxes can not work with same external IP, because asterisk only work with IP not domain name
     
  16. dicko

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    Re:mobile sip phone

    To clarify:

    If the box is behind a NAT then sip_nat.conf should not be blank or your SIP headers will not be rewritten correctly, if the headers aren't so written then there will be no audio, the contortions you do with your firewall might allow a registration, (don't put your box in the DMZ ;) ) but you will almost always screw up the audio (rtp).

    Many Elastix Boxes can share the same IP, they just cannot share the same SIP registration Ports. a domainname will resolve to an IP address if two names resolve to the same address and they use the same port you are technically fuc%^ed. This is all just basic networking and nothing to do with Elastix or Asterisk per se. , fix your network, end of story :)


    dicko
     
  17. souxinh

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    Re:mobile sip phone

    Thanks Dicko and Siu for input.

    I only on elastix box and using one dynamic ip sharing with other windows system. (IIS) I think that, in order for me to be able to register softphone from anywhere OUTSIDE of my network, like my friend house or at my work place, then I need static ip. Because currently, I have 2 local network (one WAN dynamic IP, and 2 routers each with it own local net (192.168.2.x and 192.168.3.x) and all my softphone (3 on .3.x and 2 on .2.x) are registed with my elastix fine (elastix on .3.x).

    I think at the moment, having static IP will solve my problem. I will post the result once I get static IP at my home!;)

    Thank you all of you

    Souxinh
     

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