[Asterisk-Users] Getting a Cisco gateway to work with Asterisk

[Mark Dutton]

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 asterisk at home, 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:390 at dev.datamerge.local:5060 SIP/2.0 
Via: SIP/2.0/UDP  192.168.44.23:5060;branch=z9hG4bK3016D6 
From: <sip:894742460 at dev.datamerge.local>;tag=391004-1A5E 
To: <sip:390 at dev.datamerge.local> 
Date: Sun, 22 May 2005 14:29:25 GMT 
Call-ID: BB5B196D-CA0411D9-803BE53F-D6B5D89 at 192.168.44.23 
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:894742460 at 192.168.44.23>;party=calling;screen=yes;privacy=off 
Timestamp: 1116772165 
Contact: <sip:894742460 at 192.168.44.23:5060> 
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:894742460 at 192.168.44.23:5060>

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 <blairs at isc.upenn.edu>
Subject: Re: [Asterisk-Users] Getting a Cisco gateway to work with
	Asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <4290A0E7.9010505 at isc.upenn.edu>
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:. at datamerge.local.
>  
> I cannot figure out how to get it to identify as 
> sip:cisco at datamerge.local. 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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