[Asterisk-Users] Help with Incoming calls

[Emilio Panighetti]

Hi.

I looked at some examples with Cisco gateways with FXO ports, but I 
have DIDs on ISDN lines. I don't know what I'm missing. In fact, my 
gateway can connect directly to the IP Phone's IP address and to SER, 
and I see what it seems a normal SIP message on Asterisk's debugs, then 
somehow looks like Asterisk is rejecting the incoming calls and it 
doesn't even say anything on the console with debug level 31, except 
for when I do a SIP debug, which seems normal to me except for the fact 
that Asterisk always returns "SIP/2.0 481 Call Leg Does Not Exist" to 
the gateway.

On Oct 15, 2004, at 5:40 PM, Emilio Panighetti wrote:

> 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
>
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