On Mon, Jan 4, 2010 at 10:38 AM, Eric Varsanyi <[email protected]> wrote:
> Thanks! I fixed this as you suggested and the From header is now:
>
> From: "E Varsanyi" <sip:[email protected]>;tag=4024212822987978946


I believe the From should look like :

>From : <[email protected]>

Please select the advanced section of the caller id screen and set it there.

When the 407 comes back from voip.ms, sipxbridge will answer the challenge.

Regards

Ranga

cc: sipx-dev as I think this could be of general interest.



>
> (as seen on the external interface heading toward the internet)
>
> There's still nothing in the INVITE that looks like authentication 
> information and voip.ms still returns the '407 Proxy Authentication Required' 
> which is ACK'ed by us. After our ACK no more external traffic occurs and the 
> 650 just gets a fast busy.
>
> Should sipxproxy respond to the 407 challenge or should it have put in the 
> authentication headers in the initial INVITE?
>
> Would another merged.xml with the fixed caller ID settings help?
>
> -Eric
>
> On Jan 4, 2010, at 9:14 AM, M. Ranganathan wrote:
>
>> On Sun, Jan 3, 2010 at 9:56 PM, Eric Varsanyi <[email protected]> wrote:
>>> I've followed the setup as closely as possible from here: 
>>> http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration
>>>
>>> I am registering successfully with voip.ms and I can receive inbound calls 
>>> w/o issue. I have a linux firewall (iptables, using shorewall) set to 
>>> forward my external port 5080 to my internal port 5080 on the sipx test box.
>>>
>>> I can use 'Telephone' (the google SIP client) from behind my NAT (directly, 
>>> without involving sipxecs) w/o issue to make outbound calls.
>>>
>>> I'll attach the packets in question below, but the basic issue is all my 
>>> outbound calls end up getting challenged with a '407' response from voip.ms 
>>> and then I get a fast busy on the inside extension (a polycom 650). I don't 
>>> understand how to configure the 'local' part of sipxbridge and it seems 
>>> 'wrong' that its talking about port 5090 which I haven't configured 
>>> anywhere. Its not shown in the traces below but the un-natted packets 
>>> arriving at the firewall are coming from the machine running sipxbridge, 
>>> not from the polycom phone.
>>>
>>> I'm running a recent (1/3) svn checkout of sipxecs with polycom 3.2.2 
>>> firmware.
>>>
>>> I need some hints or pointers as to where to start digging on this. It 
>>> seems like I need a way to make sipxbridge send the authentication info 
>>> with the INVITE even though I'm registering. I've configured voip.ms to 
>>> expect authentication rather than a static IP (since I don't have a static 
>>> IP in this test rig).
>>>
>>> Thanks for any clues,
>>> -Eric Varsanyi
>>
>>
>>
>>
>> Hello!
>>
>>
>> Looking at merged.xml.gz indicates that the From header is
>>
>> From: <sip:[email protected]>;tag=8A56E859-49C36EF6
>>
>>
>> This indicates that the caller-id has not been properly set.
>>
>> See :
>>
>> http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration#7._Configure_the_Caller_ID_settings
>>
>> Ranga
>>
>>
>>
>> --
>> M. Ranganathan
>
>



-- 
M. Ranganathan
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