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

Mon May 23 07:38:28 MST 2005

I tried that, but it is not working for me with Asterisk at Home v1.0 :-(

-b

> 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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