Hi all
I have a new Cisco 2801 with 4 BRI interfaces.
I have successfully configured it to work with other call managers, but the problem I have with Asterisk is that Asterisk needs all user devices to register and I can’t figure out how to get the Cisco to do this properly.
When I try to call into my Asterisk domain via the Cisco gateway, I see the error :-
Found no matching peer or user for ‘192.168.44.23:49551’
Here is the beginning of the SIP conversation. I am calling into my DID and Cisco is passing through the last three digits to Asterisk. In this case 390.
Sip read:
INVITE sip:390@dev.datamerge.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3C2156F
From: sip:894742460@dev.datamerge.local;tag=1CFBE590-1DFE
To: sip:390@dev.datamerge.local
Date: Sun, 22 May 2005 07:26:27 GMT
Call-ID: A4D41C27-C9C911D9-853190DA-82E86D41@192.168.44.23
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 2765324287-3385397721-2174091283-2141291050
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 15
Remote-Party-ID: sip:894742460@192.168.44.23;party=calling;screen=yes;privacy=off
Timestamp: 1116746787
Contact: sip:894742460@192.168.44.23:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 329
v=0
o=CiscoSystemsSIP-GW-UserAgent 777 5948 IN IP4 192.168.44.23
s=SIP Call
c=IN IP4 192.168.44.23
t=0 0
m=audio 17928 RTP/AVP 8 18 98 3 0 19
c=IN IP4 192.168.44.23
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:98 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
20 headers, 14 lines
Using latest request as basis request
Sending to 192.168.44.23 : 5060 (non-NAT)
Found no matching peer or user for '192.168.44.23:49551’
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 98
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 19
Peer audio RTP is at port 192.168.44.23:17928
Found description format PCMA
Found description format G729
Found description format GSM-EFR
Found description format GSM
Found description format PCMU
Found description format CN
SNIP…
I have tried putting the Cisco entry into sip.conf with a fixed host address, dynamic host address and username/secret pair and both to no avail.
When I use other gateways, Asterisk asks the gateway to authenticate, but in this case after the above block, it tells the Cisco to go away and nothing works.
Here is my Cisco entry in sip.conf
[cisco]
context=from-pstn
host=dynamic
type=friend
username=cisco
secret=0000
This is driving me mad.
Has anyone here successfully setup a Cisco gateway for incoming calls?
Regards
Mark