[Asterisk-Users] Getting a Cisco gateway to work with Asterisk
Alistair Cunningham
acunningham at integrics.com
Mon May 23 06:13:30 MST 2005
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 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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