[asterisk-users] Re: ast 1.2.x - cisco 7970 behind nat to external asterisk with no nat

Tue Apr 24 14:46:34 MST 2007

Here is a followup:

I've now tried SIP 7.0.5 which also doesn't work. I've also got
debugging information from both sites (1.4.2, nat, local) and (1.2.16,
no nat, remote) which I will paste below. Any help would be greatly
appreciated. It looks to me like the issue is the following:

Authorization: Digest
username="8080",realm="asterisk",uri="sip:10.0.2.10",response="f990f963433d72944ca125d5c62c275d",nonce="13a80653",algorithm=MD5
Content-Length: 0

That appears on the 1.4.2 site, but not the 1.2.16 side. Is this why
the phone isn't registering? I don't know enough about SIP to know for
sure.

SIP ON REMOTE BOX:
------------------

<-- SIP read from XXX.XXX.XXX.XXX:55511:
REGISTER sip:pbx.somedomain.com SIP/2.0
Via: SIP/2.0/UDP 10.0.2.20:5060;branch=z9hG4bKf7e4cbea
From: <sip:125 at pbx.somedomain.com>;tag=0015faa0e8cf000779e2fc93-88fdab30
To: <sip:125 at pbx.somedomain.com>
Call-ID: 0015faa0-e8cf0005-9f301cb5-e7d34d98 at 10.0.2.20
Max-Forwards: 70
Date: Tue, 24 Apr 2007  GMT
CSeq: 103 REGISTER
User-Agent: Cisco-CP7970G/8.0
Contact: <sip:125 at 10.0.2.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0015faa0e8cf>";+u.sip!model.ccm.cisco.com="30006"
Content-Length: 0
Expires: 3600

--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.0.2.20 : 5060 (NAT)
Transmitting (NAT) to XXX.XXX.XXX.XXX:55511:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.20:5060;branch=z9hG4bKf7e4cbea;received=XXX.XXX.XXX.XXX
From: <sip:125 at pbx.somedomain.com>;tag=0015faa0e8cf000779e2fc93-88fdab30
To: <sip:125 at pbx.somedomain.com>
Call-ID: 0015faa0-e8cf0005-9f301cb5-e7d34d98 at 10.0.2.20
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:125 at 216.145.22.110>
Content-Length: 0

---
Transmitting (NAT) to XXX.XXX.XXX.XXX:55511:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.2.20:5060;branch=z9hG4bKf7e4cbea;received=XXX.XXX.XXX.XXX
From: <sip:125 at pbx.somedomain.com>;tag=0015faa0e8cf000779e2fc93-88fdab30
To: <sip:125 at pbx.somedomain.com>;tag=as67521997
Call-ID: 0015faa0-e8cf0005-9f301cb5-e7d34d98 at 10.0.2.20
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1810bf00"
Content-Length: 0

SIP ON LOCAL (NO NAT) BOX:
--------------------------

<--- SIP read from 10.0.2.20:51950 --->
REGISTER sip:10.0.2.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.20:5060;branch=z9hG4bKb64f7d91
From: <sip:8080 at 10.0.2.10>;tag=0015faa0e8cf0002ce03525c-f41c3afb
To: <sip:8080 at 10.0.2.10>
Call-ID: 0015faa0-e8cf0002-ce1851de-2d1c9545 at 10.0.2.20
Max-Forwards: 70
Date: Tue, 24 Apr 2007  GMT
CSeq: 102 REGISTER
User-Agent: Cisco-CP7970G/8.0
Contact: <sip:8080 at 10.0.2.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0015faa0e8cf>";+u.sip!model.ccm.cisco.com="30006"
Authorization: Digest
username="8080",realm="asterisk",uri="sip:10.0.2.10",response="f990f963433d72944ca125d5c62c275d",nonce="13a80653",algorithm=MD5
Content-Length: 0
Expires: 3600

<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.0.2.20 : 5060 (no NAT)

<--- Transmitting (no NAT) to 10.0.2.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.20:5060;branch=z9hG4bKb64f7d91;received=10.0.2.20
From: <sip:8080 at 10.0.2.10>;tag=0015faa0e8cf0002ce03525c-f41c3afb
To: <sip:8080 at 10.0.2.10>
Call-ID: 0015faa0-e8cf0002-ce1851de-2d1c9545 at 10.0.2.20
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:8080 at 10.0.2.10>
Content-Length: 0

<------------>
pbx*CLI>
<--- Transmitting (no NAT) to 10.0.2.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.2.20:5060;branch=z9hG4bKb64f7d91;received=10.0.2.20
From: <sip:8080 at 10.0.2.10>;tag=0015faa0e8cf0002ce03525c-f41c3afb
To: <sip:8080 at 10.0.2.10>;tag=as3d34555a
Call-ID: 0015faa0-e8cf0002-ce1851de-2d1c9545 at 10.0.2.20
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:8080 at 10.0.2.20:5060;transport=udp>;expires=3600
Date: Tue, 24 Apr 2007 21:40:09 GMT
Content-Length: 0

Thanks for your help!

On 24/04/07, Matt Gibson <diwelf at gmail.com> wrote:
> Hi All,
>
> As the subject describes, has anyone gotten this to work? I am running
> an asterisk 1.2.16 server, and am trying to register my cisco 7970
> remotely to it, but it just won't go.
>
> I am running 1.4.2 internally and the phone registers fine to it. I'm
> using the latest firmware (i think) - 8.2.1S
>
> On the server in question I have tried the following for the sip declaration:
>
> qualify=never
> nat=no (yes)
> defaultip=(natip)(externalip)
> md5secret=<md5pass>
> or
> secret=<secret>
>
> Nothing seems to work, and I continually get "sip 401 unauthorized"
> messages on the console when the phone tries to register.
>
> I've spent a number of hours on this googling and searching for anyone
> working with 1.2 and 7970's, but I can't find any information. Any
> help would be much appreciated.
>
> Scenario:
>
> cisco 7970 -> switch -> pfsense/soekris/nat -> cable modem -> remote pbx
>
> Local firewall has port forwarding on for 5060 tcp/udp to my internal
> * box, and also for UDP 10000-30000 port forwarded to local * box as
> well. Is there anything else I can try?
>
> Thanks,
> Matt
>