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Thread: How to: Freedompop number with freepbx/asterisk

  1. #31
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    Quote Originally Posted by RFK View Post
    Same here: 488 Not Acceptable Here
    Enable SRTP.

  2. #32
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    I think I got the phone making calls regardless of the error message. So outgoing works (attaching pictures from MicroSIP). Incoming not working on the MicroSIP softphone yet.

    Used the advice of changing Display Name to FreedomPop Phone which possibly helped make it work.

    Siliconvalley785

    Name:  Freedompop signalwire enable OPUS and RTP Port changes.JPG
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Size:  29.0 KBName:  Freedompop signalwire settings 1.JPG
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Size:  54.9 KB

  3. #33
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    Specifying domain as zoommediaplus-outbound.dapp.signalwire.com does not work for me. Are you sure that is the settings actually used? Could you do a protocol dump? In my case auth. domain must match sip server, no other settings work. I have no problem with incoming calls and Display Name makes no difference.

  4. #34
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    Well if i tried to specify the domain as Zoommediaplus-496822d6c598.sip.signalwire.com I get Incorrect Password. But if I leave it as before I can make outgoing calls. Incoming still doesn't work, probably because it's not registering.

    If you can provide a utility or software I should use I'll do a protocol dump for you.

  5. #35
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    I downloaded MicroSIP, entered account name, username, domain as zoommediaplus-496822d6c598.sip.signalwire.com and password leaving everything else as is except for forcing TCP and SRTP and I can make outgoing calls. Caller ID is some other number though. Incoming calls don't work simply because MicroSIP does not send REGISTER packet, at least I don't see it. Please try Linphone since it works for me, at least Linux version and I can see REGISTER on the wire.

  6. #36
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    Quote Originally Posted by siliconvalley78 View Post
    I think I got the phone making calls regardless of the error message. So outgoing works (attaching pictures from MicroSIP). Incoming not working on the MicroSIP softphone yet.
    Outgoing calls work with the configuration you provided and caller id is correct. For incoming calls please see my previous message. It seems like one may need two clients, one for incoming calls and another for outgoing if correct caller id is required.

  7. #37
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    Did more work this morning and was able to get incoming and outgoing working on Linphone for Windows.
    Name:  linphone 1.jpg
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Size:  17.3 KB This picture is easy.
    Name:  linphone 2.JPG
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Size:  42.1 KB This picture isn't easy.
    for Sip Address this is sip:[email protected]
    for Sip Server Address this is <sip:zoommediaplus-xxxxxxxxxxxx.sip.signalwire.com;transport=tls>
    For Route this is sip:zoommediaplus-outbound.dapp.signalwire.com;transport=tcp
    Name:  linphone 3.JPG
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Size:  12.0 KB Just make sure IPv6 isn't selected.
    Name:  linphone 4.JPG
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Size:  39.9 KB Make sure you choose the account with the @zoommediaplus entry.

    It does show s a random number when calling from Linphone to your receipient though, as you mentioned.

  8. #38
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    No need to enter anything in Route field - I found no difference whatsoever.
    No need to bother with presence either - I haven't touched it and for the best of my knowledge signalwire server does not support it.
    Regarding very first post about asterisk adding security=yes resolves outgoing calls. I wonder what is needed for incoming, most likely change registration from UDP to TCP but I am not familiar with registration string syntax.

  9. #39
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    I don't really care much for fp's voice/text. Just the data. I was able to get the windows version of zoiper to connect and make calls. Experienced the issue with the caller id not showing up with the correct # on the recipient's end. This leads to another problem, the call doesn't show up in the freedompop.com call log. So, not sure if it's logged as a call or not.

    So, as a firm believer of K.I.S.S. I opted to do the following. Installed the fp messaging app on an old/spare rooted android phone. With the device in airplane mode all you need to enter is the phone # and password to register the app. Registered an account then did a titaniumbackup of it. Repeated for subsequent accounts. Once a month I'll go through the rigamarole of making some calls under each acct. To switch between just restore the data.

    YGWYPF!!

    Interestingly, i'm having great issues getting the lte sim to work on my old axon 7. It'll connect sometimes at hspa, rarely lte. Yet, same sim works fine in a wifi hotspot device.. Go figure. Also, no TTL bs to get full speeds (close to 70 down, 10 up mbps).

  10. #40
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    Quote Originally Posted by qtlin View Post
    No need to enter anything in Route field - I found no difference whatsoever.
    No need to bother with presence either - I haven't touched it and for the best of my knowledge signalwire server does not support it.
    Regarding very first post about asterisk adding security=yes resolves outgoing calls. I wonder what is needed for incoming, most likely change registration from UDP to TCP but I am not familiar with registration string syntax.
    correction: encryption=yes

  11. #41
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    Quote Originally Posted by GPz1100 View Post
    the call doesn't show up in the freedompop.com call log. So, not sure if it's logged as a call or not.
    Well, you have blown the cover off one of two unindented features of new FP VoIP server.

  12. #42
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    <Shrug>... It's a half @ssed system. I just want the data, don't care for the texts/calls. Maybe it's a feature...

  13. #43
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    I managed to configure Acrobits on Android to handle both outgoing and incoming calls in one account. The only problem is that for outgoing calls, CallerId on other party's phone displays random numbers from California. Any idea how to solve it?

  14. #44
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    Quote Originally Posted by qtlin View Post
    Attention all FP trunkers here:

    1. SignalWire requires SRTP
    2. Incoming calls are not routed unless TLS or TCP is used for SIP signalling.

    Question1: how to reconfigure existing FP freepbx trunk to satisfy above requirements?
    Question2: outgoing calls display some bogus caller id number. FP client submits INVITE via a different host and caller ID is correct. Is it possible to do that in freepbx?

    Protocol observation tips: Use iPhone client, it does SIP over TCP. Android uses TLS and simple ngrep is useless.
    My home router is a linux box and I just do something like:

    Code:
    # ngrep -d internal_interface_name -W byline "" host iPhone_IP
    Then observe incoming and outgoing calls.
    I was able to successfully get a trunk working in Asterisk using encryption=yes and transport=tcp, but also seeing the random outgoing CID. Tried all sorts of variations of using zoommediaplus-outbound.dapp.signalwire.com as the host, proxy, outbound proxy, but so far unable to get the correct outgoing CID.

  15. #45
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    Quote Originally Posted by gadgetboyj View Post
    I was able to successfully get a trunk working in Asterisk using encryption=yes and transport=tcp, but also seeing the random outgoing CID. Tried all sorts of variations of using zoommediaplus-outbound.dapp.signalwire.com as the host, proxy, outbound proxy, but so far unable to get the correct outgoing CID.
    Configure outgoing trunk using settings as on #32, parameter with asterisks is not needed.

    Here is mine, I suspect many lines are not needed but I have very limited syntax knowledge.

    Code:
    videosupport=no
    username=yourfp11digitsnumber
    type=peer
    transport=tcp
    qualifyfreq=240
    qualify=yes
    nat=yes
    insecure=port,invite
    host=zoommediaplus-outbound.dapp.signalwire.com
    fromuser=yourfp11digitsnumber
    fromdomain=zoommediaplus-outbound.dapp.signalwire.com
    encryption=yes
    domain=zoommediaplus-outbound.dapp.signalwire.com
    disallowed_methods=UPDATE
    directmedia=no
    context=from-pstn
    canreinvite=no

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