Problem with remote extension in Netherlands

Discussion in 'General' started by wiseoldowl, Apr 14, 2009.

  1. wiseoldowl

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    EDIT: Turned out they had two routers chained together - that was the problem (see my next message).

    We have a home based Elastix/FreePBX box and recently a friend that has an extension off our system, using a Linksys PAP2 as the endpoint, asked if there was any way he could use it to connect to a friend's daughter currently residing in the Netherlands. I suggested that maybe if my friend's friend's daughter had a PAP2 she could connect to the sever as well. So he got another unit and configured it exactly the same as the one he uses (which works great) except on different extensions. We tested it, made calls back and forth, and it still worked great. He took it to his friend's home (the father of said daughter) and it worked great from there as well.

    The father then took the adapter to the Netherlands, and that's where the trouble started. It connects to the server just fine, the adapter shows up when you do "sip show peers" from the CLI, and has low ping time, you can pick up the phone and get dial tone and make another extension ring, or you can call her extension and it will ring. What we don't get is audio in either direction. Bear in mind, this is the same unit that worked great here in the U.S. with the same settings.

    So I began asking some questions and discovered that the young lady subscribes to some kind of triple play service, and I believe the company is called Tele2. Her IP address (in the range 87.209.64.0 - 87.209.127.255) resolves to "Versatel Consumer", which according to their whois, "is one of the largest ISP's in the Netherlands", and if you try to go to their web site you are redirected to a Tele2 page. The DSL modem is a Davolink DV2020, a Korean-made unit that appears designed for triple-play service. And I'm told that she does in fact get her television, telephone, and Internet from this company.

    I'm just wondering if anyone else has run into this and if so, were you able to resolve the problem? I've tried a couple obvious things (like having them set the the PAP2 to use sip ports 5061 and 5062 instead of 5060, just in case this company's phone service used 5060 - and yes I did also make that change in the extension configuration in FreePBX) and also tinkered with the RTP port min and max settings in the adapter (within the 10000-20000 range) but none of that helped. I'm just wondering if anyone else has ever tried to set up a remote extension with this particular provider and might have any hints, or in particular, might know what ports that device reserves for its own phone service. Skype calls get through just fine, so I'm not really sure why we can't get audio to or from the PAP2, but I suspect it has to do with the DV2020 (or maybe the company itself?) interfering with the audio packet stream in some way.

    I'm really at a bit of a loss here, and it doesn't help that I don't speak a word of Dutch and that the Tele2 web site is entirely in Dutch. If anyone has any suggestions on how to get the audio to work, I'd be very appreciative! The daughter's father (i.e. the friend of my friend) is visiting her today, so I'm hoping someone can tell me something useful today.
     
  2. ramoncio

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    If you see the peer registered in asterisk, then the sip port is working ok.
    If you can't hear the voice, then your problem is with the rtp traffic.
    Maybe the telco is using or blocking those ports.

    You can change the rtp ports in asterisk

    In file /etc/asterisk/rtp.conf change values for
    rtpstart=5000
    rtpend=31000

    (You can try with other ports too)

    And restart asterisk.

    After that, you will have to open in both routers this port range.
    You will have to reditect (using NAT) your rtp traffic to your Elastix box ip, and she will have to redirect her rtp traffic to her SPA's ip.

    You can also see a rtp debug in your asterisk console with

    rtp debug
    or
    rtp debug ip [remote ip]
     
  3. wiseoldowl

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    Thanks. It turns out they had two routers chained together - apparently the wireless on the telco-provided unit is so bad that it won't even get out of the same room, so they bought a Belkin F5D8233-4v3 and plugged it into the telco unit, then plugged both computer and VoIP adapter into THAT. When the VoIP adapter is connected directly to the telco unit it works fine. When plugged into the Belkin router we get no audio, however if the firewall is disabled the we get one-way audio (the person they call can hear them, but they get nothing). We're still working on possible solutions but he had to go to dinner, so the project kind of got hung up in mid-stream.

    (By the way, I now have a bit more sympathy for you guys who work tech support and try to get a customer to do something, and they want to do everything except the one thing you really want them to do. He was having trouble understanding the Dutch menus on the computer and was trying to change the language to English, and I had a page in front of me that explained how to do it (in English), and he had a nephew standing right there that could translate from English to Dutch. But do you think I could get him to hand the phone to the nephew so the nephew could understand what to do, instead of (often incorrectly) relaying what he thought I was saying to the nephew, then asking the nephew "What does this say?" "Well, what does THIS say?" - and mind you this had nothing to do with the problem, and the Belkin router menus were in English. Aaarrrgh! He was at least smart enough to know that Vista is "an instrument of the devil", though (his words) so I couldn't be too upset with him). :)
     
  4. rollinsolo

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    http://www.elastix.org/index.php?option ... t=10#20732

    My forum question was very similar, my remote extensions are not in the Netherlands, (wish they were), but I am having the same problems with audio, not sure if the link above works but its 20732 "Please Help my cannot dial remote extensions.


    I got to the point of being able to dial the remote extensions and the remote extension rings but when answered neither party can hear and the calling party continues to hear the ring then into voicemail, I only got this far by changing qualify to no in the extensions page of elastix. Now they show up in sip show peers as unmonitored when I assume in a perfect world they would show up like the others as OK (44 ms) etc. Hopefully we are working on the same issue and can resolve it together.

    Name/username Host Dyn Nat ACL Port Status
    6900/6900 externalIP D N 5060 Unmonitored
    7453/7453 externalIP D N 5060 Unmonitored
    7452/7452 externalIP D N 53064 Unmonitored
    7450/7450 externalIP D N 63830 Unmonitored
    7449/7449 externalIP D N 5060 Unmonitored
    7447/7447 192.168.1.10 D N 5060 OK (12 ms)
    7446/7446 192.168.1.9 D N 5060 OK (11 ms)
    7445/7445 192.168.1.2 D N 5060 OK (10 ms)
    7444/7444 192.168.1.3 D N 5060 OK (10 ms)
    7443/7443 192.168.1.5 D N 5060 OK (11 ms)
    7442/7442 192.168.1.7 D N 5060 OK (11 ms)
    7441/7441 192.168.1.8 D N 5060 OK (7 ms)
    7440/7440 192.168.1.11 D N 5060 OK (112 ms)
     
  5. dicko

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    Are more than one remote extension behind any external firewall? or are all those "externalIP" different?
     
  6. rollinsolo

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    yes they are different ip's at 5 different locations
     
  7. dicko

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    7452 and 7450 seem to be using NAT translation at the far end, can they be "qualified"
    the other remotes seem to be port translations. Are these routers set to port forward?, have you tried without port forwarding? What are these remote devices?,

    I ask all this as many ATA's/phones have nat settings, and then you have in effect a nat behind a nat and things can get squirrely.

    depending on the nature of the nat on the far end router (and your own) it becomes complicated.
    I suggest you choose one appropriate IP address and then "sip debug ip <that address> to see why it is not "registering", Once registered then as romoncio said, "rtp debug" and "rtp debug off" will reveal the
    ip and port being used for the negotiated RTP session (audio), you will see (hopefully) 2 streams one in and one out, where "len" is the packet load.
    Empirically I have found that port forwarding at the far end router is often unnecessary as once the phone has negotiated a connection with the asterisk box (assuming it has been set up to "register" (an important setting in the ATA/phone), a modern router/firewall will do all the port translation necessary
     
  8. rollinsolo

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    Let me further clarify, yes they are remote extensions at 5 locations, they are behind NAT firewalls, and yes I changed all of the external ip's from the actual different IP addresses. I have configured my sip_nat.conf file to use the static IP that the pbx is tied to and all of these external SIP phones can dial out to any regular phone # or any extension that is local to the pbx, they can dial other remote extensions but the same result occurs as inbound to these remote's that the caller just hears 4 rings then voicemail and the receiving party only hears dead air. Any help is much appreciated.

    Here is my sip_nat.conf file I have changed it many times then do amportal restart just to make sure but to no avail.

    nat=yes
    externhost=pbxip.mydydns.com
    externrefresh=60
    localnet=192.168.1.0/24
     
  9. rollinsolo

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    I will try that now, and post a pastebin of the results, thank you. I took all port forwarding off at the remote sites.
     
  10. dicko

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    I think I understand the scenario.
    the last bit of my post was to debug the registration first (then the audio) the fact that they can call out suggests that your asterisk box natting is correct the fact that you can't call them suggests that the far end firewalls or the phones themselves are miss configured, so from asterisk CLI

    sip debug ip <choose one>

    will show every few minutes attempts to register the phone with the device, the phone sends a packet and the asterisk box responds, it tries a few times and gives up, you will see these packets and where the are being sent (it should be to a resolvable IP address derived from the VIA address, which should be the external ip of the far end router, I hope you are following me so far, until the phones register then there will be no route to get to the phone to make it ring (well it can blindly send an invite packete to the IP/Port it knows about but it probably wont work or the registartion process would have succeeded) , if the far end router is mis-configured then . . . .

    over to you. . .
     
  11. wiseoldowl

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    I can't speak for anyone else but I have never been able to get externhost to work. This is what we use (with specific details xxx'd out):

    nat=yes
    externip=xxx.xxx.xxx.xxx
    fromdomain=xxx.xxx.com
    localnet=192.168.0.0/255.255.255.0

    Unfortunately this causes external sip connections to fail if the external IP address changes, so we run a perl script and a cron job to address that issue.
     
  12. rollinsolo

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    hmmm the only thing is the remote phones are ringing, and they are showing caller ID, I was at a remote site and tested all scenarios of in and outbound calls. I can try to port forward udp 10000-20000 and see if that makes a difference, but it is so close, I know its just a simple setting away to make it right.
     
  13. rollinsolo

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    dyndns.com is there for any changes to a dynamic ip, I am using dydns with a static IP for ease of use of the name, but if you only have one address then it is a free service you should look into it and take out the script. Not sure if you have already tried that, I have never seen your config but I am sure there are many.
     
  14. dicko

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    why not and now is the time to debug the rtp streams to see if they are biderectional. . .
     
  15. rollinsolo

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    Ok here are the debug results

    http://pastebin.com/m7de2ab6f

    I am not that good at reading these yet. I know 200 ok means registered but from there ugg.
     
  16. rollinsolo

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    and here are the RTF feedback from a call to my remote extension

    Got RTP packet from 173.78.144.2:10070 (type 00, seq 006890, ts 674256760, len 000160)
    Got RTP packet from 173.78.144.2:10070 (type 00, seq 006891, ts 674256920, len 000160)
    Got RTP packet from 173.78.144.2:10070 (type 00, seq 006892, ts 674257080, len 000160)
    Got RTP packet from 173.78.144.2:10070 (type 00, seq 006893, ts 674257240, len 000160)
     
  17. dicko

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    Well the NOTIFY's (keep alive) seem to be working, do you have qualify(register) on on both the phone and the extension?
     
  18. rollinsolo

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    If I change qualify to yes in elastix then the calls go straight to VM, the phones (Atcom 510) have a enable subscribe is that the same thing?
     
  19. rollinsolo

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    The phone interfaces all show registered.
     
  20. blackgecko

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    just as a reference, can you do a extended ping from the pbx to the remote extensions that ara having problems, just to see what the results are like
     

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