[Asterisk-Users] Grandstream problem

[William Carlson]

Does everything work fine now? I am still having problems with SayUnixTime. Voicemailmain2 works though. The one simple AGI script I wrote doesn't do anything. Asterisk starts playing and the grandstream just rings. Both work fine on other channels/sip phones.
   Thanks,
      Will
   


  ----- Original Message ----- 
  From: Wim Venneman 
  To: asterisk-users at lists.digium.com 
  Sent: Friday, November 07, 2003 1:46 PM
  Subject: Re: [Asterisk-Users] Grandstream problem


  Thanks William,

  Works fine now.

  Wim
    ----- Original Message ----- 
    From: William Carlson 
    To: asterisk-users at lists.digium.com 
    Sent: Thursday, November 06, 2003 9:43 PM
    Subject: Re: [Asterisk-Users] Grandstream problem


    try 
    disallow=all
    allow=ulaw

    under the general section of sip.conf

    that half fixes it for me calls between phones work but talking to asterisk has some problems.
      ----- Original Message ----- 
      From: Wim Venneman 
      To: asterisk-users at lists.digium.com 
      Sent: Thursday, November 06, 2003 2:29 PM
      Subject: [Asterisk-Users] Grandstream problem


      Hi,

      I installed Asterisk an all works fine exept for Grandstream.
      When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok
      When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so)
      It's the same when I call between two Grandstream phone's. Call from phone1 to phone 2, I pick up phone2 and afther 3 seconds I get congestion tone from both phone's.

      Info from command *CLI>
      -- Executing Dial("SIP/phone2-a030a", "sip/phone1") in new stack
      -- Called phone1
      -- SIP/phone1-663a is ringing
      -- SIP/phone1-663a answered SIP/phone2-a030a
      -- Attempting native bridge of SIP/phone2-a030a and SIP/phone1-663a
      == Spawn extension (sip, 1,1)  exited non-zero on 'SIP/phone2-a030a'

      and I get congestion

      Can anyone give me a direction to solve my problem?
      Thanks in advance,

      Wim
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