Question

Cisco 7960 phones and Trixbox, both behind a NAT/Firewall. Lines will not register on Proxy.

Asked by: jsilberberg


Having some issues with the NAT/Firewall settings in the SIPmacaddress.cnf configuration files for Cisco 7960 phones. The inside phone next to the Box can register and work with our Trixbox and register a second line with a Proxy outside of the NAT/Firewall.  But an outside Phone on a DSL also NATed can only get the TFTP to work and phone to load.  However the Lines will not register and come active to any of the Proxies..

I believe that the issue is in this paragraph..  But I can't find any details on the macros at all, or specifically what values should be supplied in the nat_address and nat_received_processing.

Is it the Servers NAT, the Phones NAT the Servers Public or the Phones Public IP that is supposed to be in the address field ???

# NAT/Firewall Traversal
nat_enable: "1"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16384"
end_media_port:  "32766"
nat_received_processing: "-default-"

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Asked On
2007-02-06 at 10:05:19ID22151699
Tags

7960

,

trixbox

,

cisco

Topic

Voice Over IP

Participating Experts
2
Points
125
Comments
10

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Answers

 

by: rpalmeira22Posted on 2007-02-07 at 10:25:55ID: 18487215

is the proxy behind the NAT/Firewall box as well? if that's the case then what you have is

Phone(outside)--->DSL--->Internet cloud->NAT/Firewall<---proxy<---Phone(inside)

which won't work unless the proxy is basically public. What are you doing for far end NAT traversal? This is typically the job of the proxy/sbc/stun server whatever. which requires that device to be public.

 

by: jsilberbergPosted on 2007-02-07 at 11:23:49ID: 18487665


Well It looks more like this ....  An the Term Proxy is probably where I have a comprehension issue..


Phone(outside)--->DSL--->Internet cloud->NAT/Firewall<--ASTERISK(inside)
                                                |                                             |
        SIP Trunk Provider  ------->|                                             |--<--> Phones(inside)

Howwould I  configure a proxy/sbc/stun server for CISCO 79nnPhones ?

TIA,
JMS..

 

by: FrabblePosted on 2007-02-11 at 19:28:00ID: 18512638

Proxy is the host the SIP client registers with and uses to set up calls.
Since SIP passes addresses and ports in the headers, with NAT and your client is using a private address, you need to provide the phone's WAN address - this is the nat_address entry. Otherwise you would use an outbound proxy which will change the address in the headers transparently and will be used for registration/calls.

Full Cisco reference here:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a00800d6a6d.html#1022331

To get SIP working properly with this situation, you'll also need to redirect the ports used for control and media. 12 media ports is enough for 6 lines, so set the end_media_port to 16395. On the NAT DSL device, redirect UDP 5060 and range UDP 16384 - 16395 to the phones private IP address.

That's it.

 

by: rpalmeira22Posted on 2007-02-12 at 18:13:36ID: 18520094

sorry, long weekend.
ok, see something like http://www.vovida.org/pipermail/vocal/2002-August/006435.html for how you would configure it with STUN.

there are a couple different options here depends on how you want to approach it. one would be to put a VPN in place and effectively make it a LAN. you'll still have some issues here. another options would be to use a different STUN or NAT device or a public SBC to handle the devices then have the phones all register there instead of the Aterisk box inside the network.

 

by: jsilberbergPosted on 2007-02-22 at 12:13:24ID: 18590999

Sorry for the Delay - Been side tracked hope to get back on this early next week ---

 

by: rpalmeira22Posted on 2007-02-22 at 13:36:20ID: 18591773

no prob. getting side-tracked is nothing new :) just post back with an update when you get a chance.

cheers,
-rp

 

by: jsilberbergPosted on 2007-02-25 at 18:01:49ID: 18606944

Making lotrs of progress behind the FW and connecting to the VoIP provider through the FW.
Trix likes to change my settings for my when I up things like sip_additional.cnf in a vi session.

Will start back on getting the outside 7960s up now.  

Side note, in the CISCO Manuals it talks about Speed_Labeln and Speed_Dialn to assign the Line / Speed Dial keys values, but then it had a NOTE  "You cannot use this parameter in the configuration file." Anyone know how you do specify a value ??  

http://cisco.com/application/pdf/en/us/guest/products/ps2156/c2001/ccmigration_09186a00801d1972.pdf  Page 167..

TIA,,,


 

by: rpalmeira22Posted on 2007-02-26 at 18:53:49ID: 18614395

you should be able to use speed_lineX and speed_labelX in config file see section 5-18 in the admin guide that you link to. I haven't used it much in configs, usually just let individual users set as necessary via menus. But it should be able to handle in the config file.

 

by: jsilberbergPosted on 2007-02-28 at 08:52:02ID: 18626614

Well,

   I can't get a few things working in the 7960(s) yet.

  The SpeedLine/Dial only seems to work when set manually at the station.  Did not find your reference in the manual either.  

   Can't seem to get SIPnnnnnnnnnn.cnf to allow me to load alternate lines, Say 2 4 6.  As soon as I skip a line it's as if the code stops aparing the file and just marks everythings as unprovisioned.

   Last issue is Intercom / paging.  Set Intercom Line as auto answer, but then a call coming in got routed to the same extension so it answered, even though it was not an "intercom" call.  Guess I will have to build an alternate group of extensions for this function :-(

   As far as the STUN/NAT issues go. Well maybe next week..

   JMS..  

 

by: jsilberbergPosted on 2007-03-13 at 11:42:35ID: 18712833

Update --

  Still fighting with Double  NAT issues but things are getting better, and a VPN tunnel solves many sins :-)

Cisco 7960 SIP 6.3  the AuoAnswer: "32"  sets the dispaly option to that Line name, but does not enable Auto Answer you still have to go to Call Preferences and tell it with line to Auto_answer..

Still oooking for a way to aletr Softkey behavior.  Would like XFER and Blind XFer to be moved up to the top menu when answering an incoming call instead of having to press more..  

TIA..
JMS..

20120131-EE-VQP-002

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