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Thread: How to: Freedompop number with freepbx/asterisk

  1. #16
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    Unhappy Freedompop changed credentials format, FreePBX does not work anymore.

    New format is using signalwire domain and authentication name is just an 11 digits phone number.
    I followed above instructions to define FreeBPX trunk. Registration is successful but attempt to make outgoing call fails with
    SIP/2.0 488 Not Acceptable Here
    Interestingly linphone 3.6.1 also can't call out any more for the same reason.
    Grandstream HT812 appears to be able to make and receive calls but outgoing caller ID is not set to my number.
    Any SIP experts out here to figure out how to use new Freedompop SIP server?

  2. #17
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    I haven't touched the fp sip settings in ages.. still works with the old layered.net entries.

  3. #18
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    Quote Originally Posted by GPz1100 View Post
    I haven't touched the fp sip settings in ages.. still works with the old layered.net entries.
    Also no change for me

  4. #19
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    Guess we spoke too soon. All 3 trunks are failing.

    I'm seeing this in asterisk -rvv

    Code:
      == Everyone is busy/congested at this time (1:1/0/0)

  5. #20
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    I can't make a call with my phone's built in SIP client. Been using it for 2 years now, no issues, until the middle of this week. I get a busy signal when dialing out.

    I tried to install the FreedomPop app on another Android phone but it doesn't seen to work. I can login fine and it sees my account details on WiFi. When I try making a call using the app, it brings up the native dialer and can't connect. I've tested the app once before when getting the credentials. Reviews seem to suggest calling on the app is broken as well, but text messages work.

    Cell data seems to be working fine though.

  6. #21
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    I was able to make some calls tonight. It does indeed look like the sip creds have changed. Same goes for the registration data too. How do we make this work in freepbx?

    I found this based on the domain of the new uri; https://signalwire.com/blogs/product...p-connectivity .

  7. #22
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    Calls seems to be working now in the FreedomPop app as mentioned above. Used SQLite to extract the new credentials.

    Username is now just 1 + 10-digit phone number as qtlin mentioned.
    example: 12131234567

    Password is 32 character hex number lowercase.

    The SIP server seems to be: zoommediaplus-xxxxxxxxxxxx.sip.signalwire.com
    example: zoommediaplus-123456a6b789.sip.signalwire.com


    Tried to config the native android SIP client, but calls are unable to connect. Using 11-digit dialing like before. Don't know if there's some special dialing requirement or other settings like proxy or TLS security.

  8. #23
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    They moved to SignalWire? Really? That's most interesting.

    *** Editing the next day to add:

    I'll dig around a little. I use SignalWire regularly on Asterisk and it is quirky to set up. But if FP is now supported by SW I should have a head start by cloning one of my working SW trunks and using the FP SIP credentials. TLS settings are usually involved. Obtaining the SIP server might be an issue if it is not present in what can be grabbed using SQLite because the "1g2h23j4jk45o67j45h35k3"@sip.signalwire.com string is always quite unobvious...
    Last edited by brg; 05-05-2020 at 09:46 AM.

  9. #24
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    Have obtained credentials.
    Username is the DID.
    Secret is a 32-character alphanumeric string
    Host/Domain is zoommediaplus-[12 alphanumeric characters].sip.signalwire.com

    CHANSIP reflects:

    Code:
    Name/username            Host                   Forcerport   Comedia Port     Status
    FreedomPop_Test/MY_DID   45.32.95.52            Yes          Yes     5060     OK (68 ms)
    [/SIZE]



    SIP Registration reflects:

    Code:
    Host                                                           dnsmgr  Username             Refresh    State                 
    zoommediaplus-[12 alphanumeric characters].sip.signalwire.com  Y       1MY_DID              105        Registered

    Thus far I am receiving:

    Error:
    407 Proxy Authentication Required

    488 Not Acceptable Here
    cause=88;text="INCOMPATIBLE_DESTINATION"

  10. #25
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    Same here: 488 Not Acceptable Here

  11. #26
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    I wonder if it's looking for a specific UA. Anyone tried some packet capture?

  12. #27
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    I have only been using sngrep to look at the SIP messaging. That's as far as I'm gonna go because I have no need for these FP DIDs for non-cell, Asterisk-based inbound or outbound calling or FP app calling and texting. I set it up long ago for sh*ts and giggles, but like many others, I only use the SIMs and the FP service for free ATT LTE when on the road.

  13. #28
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    I got the 488 Error when dialing out with the new server.

    Does anyone know how FreedomPop is blocking our SIP calls?

    Strangely the call does go out to my phone, but from some random number and then hangs up immediately. Seems like FreedomPop is probably checking the UserAgent. Dunno how to spoof it on an ObiTalk if that is the case.

  14. #29
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    Quote Originally Posted by BigBrother0 View Post
    I got the 488 Error when dialing out with the new server.

    Does anyone know how FreedomPop is blocking our SIP calls?

    Strangely the call does go out to my phone, but from some random number and then hangs up immediately. Seems like FreedomPop is probably checking the UserAgent. Dunno how to spoof it on an ObiTalk if that is the case.
    It could be spoofed via Asterisk; UA can be configured.

    I think an Obi UA could also be configured, but I am no expert there. Briefly looking I see, for example, under "ITSP Profile > SIP", various references to UA, including "UserAgentName".

    The question is what to set it to. I suppose that could be determined by looking at com.freedompop.phone.db. A very quick peek doesn't show me any reference to UA, but I do see "display_name" = "FreedomPop Phone." But that could be something I added myself years ago...

  15. #30
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    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.

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