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GVoice In Flames No More

Discussion in 'Help' started by ghurty, Mar 21, 2011.

  1. wardmundy Nerd Uno

    The missing priority=1 is definitely causing the problem at least for inbound calls on some VM machines. Haven't tested elsewhere yet. Here's a patch to fix the GV module.

    Log in as root and issue the following command. Running it multiple times is harmless because it won't find the existing string to fix any more.


    NOTE: This will only patch Google Voice additions AFTER the patch is applied. If you want to delete and re-add the accounts, that would work, or you can manually make the changes in /etc/asterisk/jabber.conf and restart Asterisk. New Incredible PBX 2 installs after 8 p.m. EDT tonight already have the patch applied.
  2. donnib New Member

    I am testing my Google Voice reliability by calling from one Google voice account to another and i see often that the server picks up the call but the call still keeps on ringing from the caller. This matched your description. I tried to add the 74.125.92.125 ip in the Webmin linux firewall as ACCEPT any traffic from this IP but i still see the problem. Not sure if i have added this correct. Any ideas ?
  3. donnib New Member

    I have also done this on my machine. I have installed PIAF and Incredible PBX two weeks ago. Is it enough to just run the above line ? You also mention and patch. What patch is that ?
  4. wardmundy Nerd Uno

    The sed command IS the patch.
  5. donnib New Member

    Hmm ok well then Incoming GV does not work stable. It is VERY unstable like every 2-3 call just keeps on ringing.
  6. wardmundy Nerd Uno

    Is the problem only when calling from one Google Voice number to another one? Are both numbers on the same machine? Are any of the GV numbers on OBi devices?

    Let me suggest you try a fresh install before we look at other alternatives.
  7. donnib New Member

    I have only tried from google to google. I am calling from GMail. No devices are on Obi devices. I don't have possibility to try fresh install right now since i have spent quite a while configuring this machine. Any suggestions ?
  8. wardmundy Nerd Uno

    From a different Google Voice account, correct?? Aside from that, my suggestion is to reinstall.
  9. donnib New Member

    Yes that is correct. Why should a reinstall help when the machine has been newly installed 2 weeks ago ?
  10. wardmundy Nerd Uno

    It may not. :wink5: If you're looking for 100% reliable, 100% of the time, Google Voice probably isn't your best bet.
  11. tm1000 Guru

    Fixed:

    http://www.the159.com/googlevoice/googlevoice-latest.tgz

  12. wardmundy Nerd Uno

  13. wardmundy Nerd Uno

    For those using the older gvoice command, Google Voice changed the registration methodology. Here's the fix.
  14. wardmundy Nerd Uno

    New, new new patch

    Here's an all-in-one patch for the 10/13 problems. On your broken system, log in as root and issue the following commands after kicking all users off their phones :wink5:

  15. eCase New Member

    Patch=fail?

    Hi Ward,

    Thanks to you and the PIAF team. I tried the commands you posted - did it recompile Asterisk? It looked like it did... but I wasn't paying 100% attention ;)
    Okay, so after the patch finished the update, I tried my google-voice trunks again.
    What I have done for the past few days is just put the Google-Voice 'Trunks' on the bottom of my default Outbound Routes, and I've used my Vitelity Routes (On the top).
    So, after the patch I put the GV trunks on top of the Outbound Routes again, but they still do not work...
    Here is the error message from asterisk log:
    "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks"
    Luckily, it at least will fail through to my Vitelity trunk, so I still have outbound calls.
    Ward, what was the patch supposed to fix?
    Prior to this patch, I recompiled my Asterisk to use sound codec uLaw & G729, and I deselected GSM codec. If this patch runs a recompile session, is it possible that my prior sound codecs switch will jam something in the patch process?
    Perhaps it's something else involving a codec issue?

    Thanks!
  16. pok New Member

    hey guys my gvoice stopped working on my pbx as well (outgoing calls, haven't tried incoming).. but i am still able to call from the gvoice website... its really frustrating.. has anyone found a solution to get it working again? port 5222 is forwarded through the router so nothing should be blocking. Thanks a bunch!
  17. darmock PIAF Developer

    Nope still the same. 0nly 78 days of service left anyways till they charge for it and based on what is here now it ain't worth paying for.

    If you ran update-fixes in the last 2 days and it is still broke then I would recommend you contact tm1000 in the forums and let him know. He has taken over stewardship of the the GV-Freepbx module.


    Tom
  18. billsimon Guru

    I've got a Google Voice-SIP gateway running on a cloud server. It's very reliable (still working out the bugs that keep it from being 100%, but that could be Google's own doing) and located only 2.1 ms away from voice.google.com so there's very low latency on that leg of the call. Inbound and outbound work great.

    If anyone would be interested in connecting to GV in this manner let me know. I'd need to collect a few small donations to keep the server running and pay for the bandwidth, but if you want to test things out first that would be fine.
  19. wauters New Member

    Does anyone have the google voice problem fixed? I am running the google voice module and I have tried changing the wait to 8, still really not working, I do get a call to go thru once in a while.
  20. jason_m New Member

    google's at it again.

    On my Asterisk system, the solution was to put some kind of audio before sending dtmf 1. I ended up making a sound file with about 50ms of barely audible fuzz/static. This would explain why lifting the receiver (with an open mic) would start the prompt playing. It seems Google has implemented some more attempts to foul up pbxes. Idealy a pbx or other automated means of answering calls would initialy generate no audio at all, as with the Wait commands.

    Answer
    Wait(1)
    Playback(custom/fuzz)
    SendDTMF(1)
    Wait(2)
    * start ivr or whatever *

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