View Poll Results: What do you think I should do?

Voters
8. You may not vote on this poll
  • Give up, use the PAP and deal with it.

    4 50.00%
  • Try another VOIP gateway/Soft-PBX app: (recommendations?)

    1 12.50%
  • Keep at it and tell me when you've got the bugs worked out w/Asterisk.

    2 25.00%
  • I did it and here's how! (info?)

    1 12.50%
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Thread: Asterisk help

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  1. #1

    Default Re: Asterisk help

    Quote Originally Posted by dswartz View Post
    context=custom-voipo
    Dswartz:
    I tried your config but I have never seen this context used. How did you define it?

    scott2020:
    I tried your config too and I don't know if I've taken a step forward or a step back. The line rings once, and then I get a male voice that says "The number or code that you have dialed is incorrect. Please check the number or code and try again."

    I noticed that the log goes nuts when I call it, and it looks like the call IS hitting the server:
    [Jun 17 16:10:18] --- (20 headers 16 lines) ---
    [Jun 17 16:10:18] Sending to 67.228.251.106 : 5060 (no NAT)
    [Jun 17 16:10:18] Using INVITE request as basis request - 420301830_32241846@64.156.174.74
    [Jun 17 16:10:18] Found peer 'VOIPo'
    [Jun 17 16:10:18] Found RTP audio format 0
    [Jun 17 16:10:18] Found RTP audio format 18
    [Jun 17 16:10:18] Found RTP audio format 4
    [Jun 17 16:10:18] Found RTP audio format 101
    [Jun 17 16:10:18] Peer audio RTP is at port 67.228.251.106:43066
    [Jun 17 16:10:18] Found audio description format PCMU for ID 0
    [Jun 17 16:10:18] Found audio description format G729 for ID 18
    [Jun 17 16:10:18] Found audio description format G723 for ID 4
    [Jun 17 16:10:18] Found audio description format telephone-event for ID 101
    [Jun 17 16:10:18] Capabilities: us - 0x4 (ulaw), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
    [Jun 17 16:10:18] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    [Jun 17 16:10:18] Peer audio RTP is at port 67.228.251.106:43066
    [Jun 17 16:10:18] Looking for {My Voipo #} from-sip (domain 192.168.0.14)
    [Jun 17 16:10:18]


    Another question is that I have the catchall route set to Extension 200 (softphone). Should I set this to Ring Group 600 (Ring all phones) or should I wait until I can confirm its operation first?

    I can't thank you all enough for your help with this. I am very excited at the possibility of getting this up and running.

  2. #2
    Join Date
    Mar 2007
    Posts
    478

    Default Re: Asterisk help

    oh, sorry, that is something i put in extensions_custom.conf to massage CID and such. you can make it just what others have done instead. i think the internal IP address is suspicious.

  3. #3

    Default Re: Asterisk help

    Try your BYOD credentials, they're available in vPanel.
    Navigate over to vPanel > Features > Preferences > "Softphone / BYOD".

    Toggle your softphone status to On and hit Save. Use the credentials provided.

  4. #4
    Join Date
    Feb 2007
    Location
    Michigan
    Posts
    2,220

    Default Re: Asterisk help

    I see that menu too.
    Did VOIPo change their mind again to continue BYOD?


    Using VOIPo services since February 2007
    Beta Tested the VOIPo Reseller Plan.
    A happy VOIPo Residential Customer

    Using VoIP devices since 12-2002
    Companies I've tried
    iConnectHere|Vonage|BroadvoxDirect|Vonage|Packet8| VOIPo
    VOIPo is a keeper!


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