[Asterisk-Users] chan_sip.c max number of retries

Robb Meredeth rjmeredeth at meredeth.com
Tue Jun 29 08:49:46 MST 2004


never mind, new server + upgrades on the phones sofware+ latest asterisk cvs = :)
  ----- Original Message ----- 
  From: Robb Meredeth 
  To: asterisk-users at lists.digium.com 
  Sent: Friday, June 25, 2004 6:59 PM
  Subject: Re: [Asterisk-Users] chan_sip.c max number of retries


  One thing I forgot to mention in the first post.  I also can call sip-sip on the old server but not on the new server, however I can call from sip-zap on the new.  When I dial from one sip to the other I get no ringback on the calling set and the called phone doesn't ring either.  After a short time the calling set gives up and goes to voicemail.  I'm sure I'm doing something dumb or I've screwed up a config I'm not thinking of right now.  I've googled and deja searches, but I'm stumped.  I even reloaded the os on the new machine and I get the exact same error.  

  Thanks again,
  Robb 



    Has anyone who's gotten this message managed to figure it out and fix it?  I've been scouring the mailing list for clues but I'm still no closer.  

    I have 2 asterisk servers, old and new.  I'm trying to  switch to the new server.  I am using a 2.4 kernel on the new and a 2.6 on the old.  I am running 0.9.0 on the old and my sip phones work fine, on the new Ihave tried 0.9.0, 0.9.1, and current CVS builds.  I never see this error on th eold machine and always on the new machine.  I have copied all my configs directly and they are plugged into the same switch as each other and the phones.   the only difference (aside from hardware) is the kernels (I had to drop back to get zaptel to compile) and the new has an X100p fxo card.    The only change I have made to the phones (zultys 4x4's) is to change the tftp server and the SIP Proxy.  If anyone can give me a direction to look in or any advice at all, I would appreciate it greatly.

    Thanks!
    Robb
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