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
<[email protected]>
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
-----------------------------------------------------------------------
-
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
