[OpenSIPS-Users] 404 Not Here

[brad smith]

Bogdan,
Thanks for responding.
I am using vitelity for my upstream; I will send them a ticket.  If they
fail to act, do you have any suggestions...switch carriers? any
config change?

Thanks again,
Brad


On Tue, Feb 26, 2013 at 7:58 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote:

> **
> Hi Brad,
>
> Thinks are a bit more complicated, it seems....
>
> In the INVITE your opensips sends to 64.....93 IP, you have the Contact
> with 192.168.1.21 (priv IP of asterisk).
>
> When you receive the BYE from 64.....93 IP, the Route hdrs are ok (the 2
> hdrs added by opensips to reflect the interface exchange), but the RURI is
> wrong - it must be the contact from the INVITE you sent, but it seems to be
> the IP of your opensips - this makes opensips to do act as strict router
> and not like a loose router....and routing gets broken.
>
> So, the 64.....93 party or some other behind it, screw up the Contact in
> the your INVITE and this alters the in-dialog requests - you should check
> with the upstream guys.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 02/25/2013 04:36 PM, brad smith wrote:
>
> I just tested an outbound call (Asterisk originate) without bridging and
> get the same '404 not here' if that helps.
>
>  Thanks again,
> Brad
>
>
> On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu <vladpaiu at opensips.org> wrote:
>
>>  Hello,
>>
>> Seems the incoming BYE does not have any Route headers, and the
>> loose_route() function returns false.
>>
>> Since you have dialog support in your script, try
>>
>> 	if (has_totag()) {
>> 		# sequential request withing a dialog should
>> 		# take the path determined by record-routing
>> 		if (loose_route() || match_dialog()) {
>>
>>
>> This way you will force matching of dialog sequential requests that have
>> no Route headers.
>>
>> Best Regards,
>>
>> Vlad Paiu
>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>
>>
>> On 02/24/2013 02:57 AM, brad smith wrote:
>>
>>  Hello,
>>
>> I am currently running opensips 1.8.1 no tls. It is
>> multi-homed with a public and private address.
>> I have a asterisk
>> 1.8.19 in the lan that is connected to opensips via lan
>> address.
>>
>>
>>  *issue*
>> A caller calls in
>> and then I place an outbound call and finally bridge the two
>> calls.
>> This works as
>> expected, except when the outbound caller hangs up first the
>> BYE never gets back to Asterisk.
>> I can see the BYE
>> reach OpenSips but a '404 not here' is returned to the ISP.
>>
>>
>>
>>
>> sip trace https://gist.github.com/5009662
>>
>>
>> opensips.cfg https://gist.github.com/5009704
>>
>>
>>
>>
>>
>>
>> thanks for your time.
>>
>>
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>>
>>
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>
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