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:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.248.10.110:5060;x-route-tag="[EMAIL PROTECTED]"
From: <sip:[EMAIL PROTECTED]>;tag=34C385A4-20B7
To: <sip:[EMAIL PROTECTED]>
Date: Fri, 15 Oct 2004 21:16:51 GMT
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]"
From: <sip:[EMAIL PROTECTED]>;tag=34C385A4-20B7
To: <sip:[EMAIL PROTECTED]>;tag=as35ed49d2
Call-ID: [EMAIL PROTECTED]
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 '[EMAIL PROTECTED]'
Then on sip.conf:
[18005550000] type=friend username=18005550000 secret=18005550000 host=dynamic qualify=2000 dtmfmode=rfc2833 [EMAIL PROTECTED] 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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