Extension 1.6-12, remote Ext hear nothing.

Discussion in 'General' started by voipmart, Dec 15, 2009.

  1. voipmart

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    Dear all & supporters,

    Now, we're using Elastix 1.6-12. Everything is perfect.
    But the problem was happen when use remote extension.
    Extension was registered...ring...hear nothing both ways.
    (But everything work probably when we try to test with Asterisk 1.4 & FreePBX 2.5 build-by-hand ... is OK)

    My network:
    ADSL --- Modem ---- LAN ---- Elastix

    Any one please support us in this case. How to NAT Elastix work with remote extension?

    Look forward to hearing from any one soonest !

    Steven.
     
  2. rollinsolo

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    I have been through the ringer on this one, took a while and now I have it working consistently, 1st hopefully your phone (ata, remote extension) has a stun option if so, do a google search for public stun servers and enter one to your liking, next stun will take care of the ata's side of things and get its messages out of its nat device just fine, now you need to work on the server's router side and first make sure your sip_nat.conf file in etc/asterisk/ has your public ip information in there http://www.freepbx.org/support/document ... -extension and finally what did it for me was buying a cheap wrt54gs version 1 to 3 (any of those will work) and loading DD-WRT SP1 VOIP use this article http://www.dd-wrt.com/wiki/index.php/Siproxd and bam my remotes have not gone offline for a long time now and I do not have to get annoying calls from my customer any longer and I look smarter as a result, jk. Sip express router acts as a proxy and all sip messages pointing to it are automatically accepted (as long as they are authenticated) anyways hope it works for you. When flashing firmwares on the router read as much as you can first before flashing or buy two of them in case you mess one up. have fun.
     
  3. mbit

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    Have you forwarded port 5060?

    What is in your sip_nat.conf?
     
  4. voipmart

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    Dear friend,

    My port forwarding as follow:
    SIP: 5060 (UDP) to Elastix box
    RTP: 10001 -> 20000 (UDP) to Elastix box

    My sip_nat.conf:

    nat=yes
    externhost=ippbxnow.dyndns.org
    localnet=192.168.1.0/255.255.255.0
    externrefresh=10

    My sip_general_customer.conf:

    #include sip_nat.conf
    bindport=5060
    bindaddr=0.0.0.0
    nat=yes
    qualify=yes
    context=from-internal
    defaultexpirey=3600
    rtptimeout=60
    rtpholdtimeout=120
    useragent=Elastix
    disallow=all
    allow=alaw
    allow=ulaw
    allow=gsm

    And each Ext, i have put allow=ulaw&alaw.

    Remote Ext can register but hear nothing on both way ! :(

    Pls let me know your idea in this case...

    Regards,

    Steven.
     
  5. dicko

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    I suggest before you do ANYTHING else you secure your server:


    and every one of your 118 extensions has the same password as the extension. I know you have no SIP trunks yet but if you did . . . , and for example extension 801 has allow=all but extension 813 has nothing in allow. (so your post wasn't completely honest)

    Please be careful what info you post here, edit your post to obfuscate any external ip address or names you have posted

    ps did this call from france go through?

    31. 2009-12-16 18:54:54 SIP/62.147... 103 "Bureau" <103> 910 ANSWERED 01:25

    p.s. I wear a white hat and didn't change anything, some where are Black Hats. . .
    and sip_general_customer.conf I hope is a typo.
     
  6. voipmart

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    Hi dicko,

    Thank for your recommendation, i will review and check careful.
    Yes, i was talk with Bureau before.

    Do u have any idea in my case ... ! Still can not now !

    Any idea will be our highest appreciate !

    Thank you !

    Steven.
     
  7. dicko

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    So you can talk to Bureau(103) in France , with audio? on extension 910?
     
  8. voipmart

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    Hi,

    He was talk with me with gsm codec, but he made trunk between 02 box and call to me on 910 ext.
    But, we need to NAT this box and remote ext can work probably.
    Pls see my above sip.conf and sip_general_customer.conf, is that correct ?
    My 910 ext : allow=alaw&ulaw&gsm and online now.
    Could u support and test call with me at this time ?

    Thank for your kindly support ! Steven.
     
  9. dicko

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    Then YOUR NAT is working (you talked to him) I suggest the PNAT box(es) between the remote extensions and your box is(are) not.


    sip_general_customer.conf is not included anywhere in sip.conf
    sip_general_custom.conf is

    I further suggest you spend more time with the nuts and bolts of VOIP/Networking setup and security than your re-branding efforts ;)
     
  10. voipmart

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    Dear dicko,

    Do u have any IM nick, Is possible for you to talk with me at this time?
    My IM:
    Yahoo: voipmart
    Skype: voipmart.vn

    Regards,

    Steven.
     
  11. dicko

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    Sorry my friend, I won't do that, it's aprivacy thing. You will have to let it all hang out for the forum to see.
     
  12. voipmart

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    Dear dicko,

    Thank you. I will re-check again carefully one more time.
    Anyway, thank for your kindly support.

    Have a good weekend !

    Steven.
     
  13. dicko

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    Et tu aussi, bon chance.
     

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