Mark,
Try writing the sip.conf stanza as:
[192.168.44.23]
context=from-pstn
host=192.168.44.23
type=friend
insecure=very
The 'insecure=very' allows any calls from this IP address to match.
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Mark Dutton wrote:
Thanks Steve
I realised the other day that I don't want the Cisco to register with
credentials. There is in fact a hidden credentials command in 12.3(8)T.
What I did was take away all registration commands from my sip-ua block in
the Cisco.
I am using [EMAIL PROTECTED], so I have created a trunk through AMP. I have
changed the settings in outbound trunk to the following and created an empty
inbound trunk on the web page with no parameters.
The result is that in Asterisk sip_additional.conf I have this block
[cisco]
context=from-pstn
host=192.168.44.23
type=friend
Now when I try to call into my gateway from the PSTN, I get the following
line immediately after the Cisco does an invite
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.23:5060;branch=z9hG4bK3016D6
From: <sip:[EMAIL PROTECTED]>;tag=391004-1A5E
To: <sip:[EMAIL PROTECTED]>
Date: Sun, 22 May 2005 14:29:25 GMT
Call-ID: [EMAIL PROTECTED]
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 3143229573-3389264345-2148466707-2141291050
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:[EMAIL PROTECTED]>;party=calling;screen=yes;privacy=off
Timestamp: 1116772165
Contact: <sip:[EMAIL PROTECTED]:5060>
Expires: 180 Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 328
v=0
o=CiscoSystemsSIP-GW-UserAgent 23 4058 IN IP4 192.168.44.23
s=SIP Call
c=IN IP4 192.168.44.23 t=0 0
m=audio 17780 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 '192.168.44.23:57704'
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 192.168.44.23:17780
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
Capabilities: us - 0x508 (alaw|g729|ilbc), peer - audio=0x10e
(gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x2 (gsm), combined - 0x0
(nothing)
Looking for 390 in from-sip-external
list_route: hop: <sip:[EMAIL PROTECTED]:5060>
You can see the line
Found no matching peer or user for '192.168.44.23:57704'
OK, now if I go into the parameters for my trunk and add the line
Port=57704
It works!!!
Problem is, the port changes. The question then is, where in my Cisco config
can I specify the listening (or return) port to 5060 so it does not pick an
arbitrary port from the pool?
Regards
Mark
Date: Sun, 22 May 2005 11:10:31 -0400
From: Steve Blair <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with
Asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
When you say identify I presume you are trying to get the Cisco to
register as a user. To the best of my knowledge it cannot do this. Instead
define a peer in sip.conf which is the gateway and place traffic matching
this peer into a context that is defined in your extensions.conf file. The
Cisco will need dial-peer statements to match inbound dialed digits and
forward all matching calls to your Asterisk box.
Mark Dutton wrote:
Can anyone please help me with sample IOS commands to get a Cisco
gateway working properly with Asterisk.
I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk.
The Cisco identifies itself as sip:[EMAIL PROTECTED]
I cannot figure out how to get it to identify as
sip:[EMAIL PROTECTED] The gateway works with other SIP servers
that don't require authentication, but Asterisk wants it to
authenticate, or at least idenitify itself and I cannot work this bit out.
If I put in the host address in my sip.conf, I still get a "cannot
find host 192.168.44.23:<random port number>, where <random port
number> is actually some random port number.
I am at my wits end.
Regards
Mark
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