[Asterisk-Users] Help with Incoming calls

[Emilio Panighetti]

Hello,

I need to make DID numbers work, and I can't seem to figure it out:

Here's what I get from a SIP debug from the Asterisk console:

Sip read:
CANCEL sip:18005550000 at 10.248.10.239:5060 SIP/2.0
Via: SIP/2.0/UDP  
10.248.10.110:5060;x-route-tag="cidpstngw1 at 10.248.10.110"
From: <sip:6175551212 at 10.248.10.110>;tag=34C385A4-20B7
To: <sip:18005550000 at 10.248.10.239>
Date: Fri, 15 Oct 2004 21:16:51 GMT
Call-ID: 59702A57-1E2611D9-89ECA2DF-D804FDDE at 10.248.10.110
CSeq: 101 CANCEL
Max-Forwards: 5
Timestamp: 1097875013
Content-Length: 0


10 headers, 0 lines
Sending to 216.52.166.110 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 481 Call Leg Does Not Exist
Via: SIP/2.0/UDP  
10.248.10.110:5060;x-route-tag="cid:pstngw1 at 216.52.166.110"
From: <sip:6175551212 at 10.248.10.110>;tag=34C385A4-20B7
To: <sip:18005550000 at 10.248.10.239>;tag=as35ed49d2
Call-ID: 59702A57-1E2611D9-89ECA2DF-D804FDDE at 10.248.10.110
CSeq: 101 CANCEL
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


  to 10.248.10.110:5060
Destroying call '59702A57-1E2611D9-89ECA2DF-D804FDDE at 10.248.10.110'


Then on sip.conf:

[18005550000]
type=friend
username=18005550000
secret=18005550000
host=dynamic
qualify=2000
dtmfmode=rfc2833
mailbox=18005550000 at default
context=from-sip
canreinvite=no
incominglimit=2
callerid=Test SIPUA <18005550000>
nat=no
disallow=all
allow=ulaw
allow=alaw

[INCOMING]
type=user
host=10.248.10.110
dtmfmode=rfc2833
canreinvite=no
nat=no
qualify=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw

On extensions.cfg:

[from-sip]
exten => 18005550000,1,Macro(stdexten,18005550000,SIP/18005550000)

That's all the significant config. I have more extensions, and they all 
can call one another, and make outgoing calls, but the calls fail 
without any indication.

Thanks