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@10.248.10.239:5060 SIP/2.0
> Via: SIP/2.0/UDP
> 10.248.10.110:5060;x-route-tag="cidpstngw1@10.248.10.110"
> From: <sip:6175551212@10.248.10.110>;tag=34C385A4-20B7
> To: <sip:18005550000@10.248.10.239>
> Date: Fri, 15 Oct 2004 21:16:51 GMT
> Call-ID: 59702A57-1E2611D9-89ECA2DF-D804FDDE@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@216.52.166.110"
> From: <sip:6175551212@10.248.10.110>;tag=34C385A4-20B7
> To: <sip:18005550000@10.248.10.239>;tag=as35ed49d2
> Call-ID: 59702A57-1E2611D9-89ECA2DF-D804FDDE@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@10.248.10.110'
>
>
> Then on sip.conf:
>
> [18005550000]
> type=friend
> username=18005550000
> secret=18005550000
> host=dynamic
> qualify=2000
> dtmfmode=rfc2833
> mailbox=18005550000@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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