[OpenSIPS-Users] 302 handling

[Alex G]

well i did make some headway on this, unfortunately i had to get tricky with
it.

Even with the get redirects, it was still not placing the correct redirect
in there. As a matter of fact,  it seems like the function was not working
at all in the failure_route. My solution involved setting an avp in the
reply route becuase both the source and destination of the paceket were the
same when it was in the failure route.  So in on reply i set an avp that
then if was true in the branch route just rewrote the host port. So great I
was able to make the call path divert but when the 2 pstn endpoints actually
link, there is no sound. There seems to be rtp when i look in asterisk's
cli, but neither side is giving me audio  :(

In the branch route, i tried with and without forecrtp proxy, but no
dice....

anyone have an idea as to what might be going on?

as always any input is greatly appreciated :)


On Sun, Aug 17, 2008 at 2:16 PM, Bogdan-Andrei Iancu <bogdan at voice-system.ro
> wrote:

> Hi Alex,
>
> Actually, after the get_redirects(), you should not do a rewiteXXXX() -
> just to t_relay(); the get_redirects() already populates the new branch with
> all the information.
>
> Regards,
> Bogdan
>
>
> Ovidiu Sas wrote:
>
>> If you want to rewrite the port, you need to use the following syntax:
>> rewritehostport("XXX.XXX.XXX.XXX:ZZZZZ");
>> where ZZZZZ is the new port.
>>
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Wed, Aug 13, 2008 at 4:54 PM, Alex G <greekman0000 at gmail.com> wrote:
>>
>>
>>> unfortunately the solution is a bit vague for what I'm trying to do...
>>>
>>>
>>> in the 302 packet the contact for redirect is sip xyz at abc.abc.abc.abc
>>>
>>> failure_route[1] {
>>>    if (t_check_status("302")) {
>>>    xlog("in redirect failure $fu");
>>>     get_redirects("*:1","redirect");
>>>      rewritehostport("XXX.XXX.XXX.XXX");
>>>     t_relay();
>>>    }
>>>
>>> this should take the contact address and rewrite the host port for it
>>> relaying it to the new location right? should be an immediate invite to
>>> abc at XXX.XXX.XXX.XXX
>>>
>>> unfortunately it doesn't rewrite the host port. It merely relays directly
>>> to
>>> the contact in the 302 packet xyz at abc.abc.abc.abc
>>>
>>> any ideas would be welcome :)
>>>
>>> thanks
>>>
>>> alex
>>>
>>> On Wed, Aug 13, 2008 at 2:38 PM, Ovidiu Sas <osas at voipembedded.com>
>>> wrote:
>>>
>>>
>>>> It is all here:
>>>> http://www.opensips.org/html/uac_redirect.html#id2519995
>>>>
>>>> Regards,
>>>> Ovidiu Sas
>>>>
>>>> On Wed, Aug 13, 2008 at 2:03 PM, Alex G <greekman0000 at gmail.com> wrote:
>>>>
>>>>
>>>>> I know there was some stuff about how to handle 302s and send forward a
>>>>> new
>>>>> invite to the diversion contact on the old mailing list archives, but
>>>>> they
>>>>> are all gone now :(
>>>>>
>>>>> wondering if anyone can help me with this.....
>>>>>
>>>>> opensips -> ua -> moved -> opensips invite contact from diversion
>>>>>
>>>>>
>>>>>
>>>>> basically opensips makes an invite to locally registered uac, the uac
>>>>> redirects to an external pstn number XXX-XXX-XXXX, opensips then needs
>>>>> to
>>>>> handle the 302 and generate an invite to XXX-XXX-XXXX
>>>>>
>>>>>
>>>>> any help would be most appreciated
>>>>>
>>>>> thanks Alex
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
>
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