[Asterisk-Users] cant connect

[Bernie]

I just installed Asterisk @home 1.0 and its up and running, i added an 
extension with the web interface, but now when I try to connect with a 
sip client (x-lite) it just times out.  here is the log from x-lite 
below.  Is there any way to view a log on the asterisk side to see whats 
going on? or anybody have an idea as to why its not working?

thanks
SEND TIME: 19038812
SEND >> 170.20.41.165:5060
REGISTER sip:170.20.41.165 SIP/2.0
Via: SIP/2.0/UDP 
170.20.11.59:5060;rport;branch=z9hG4bKD170D188076045BE8656D59545FF9678
From: Bernie ;tag=2472129214
To: Bernie 
Contact: "Bernie" 
Call-ID: [EMAIL PROTECTED]
CSeq: 49490 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0

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[Asterisk-Users] Cant connect to sjphone

[Anton Krall]

Guys.. this is killing me.. I hava a laptop running sjphone and I have 2
dial cmds to connect to that laptop in different places, first, on the main
phones context like this:
 
exten => 202,1,Dial(SIP/laptop,20,m)

so each phone can call it, and it works great.
 
Now, I have anyother cmd on a different context, which is the IVR, like
this:
 
exten => 202,1,Playback(${SONIDOS}/transferringcall)
exten => 202,2,Dial(SIP/laptop,20,m)

but when somebody from the IVR tries to call ext. 202, it gets a congestion
signal and logs show this:
 
-- Called laptop
-- Started music on hold, class 'random', on SIP/casa1-5848
Feb 23 14:09:38 WARNING[3386]: chan_sip.c:752 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 102
(Critical Request)
  == No one is available to answer at this time (1:0/0/0)

Why does it work on the phone context and not on the IVR one?
Any ideas?
 
__
Anton Krall

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