[OpenSIPS-Users] convert 180 to 183 after the fact

Jeff Pyle jpyle at fidelityvoice.com
Wed Sep 25 03:05:06 CEST 2013


No takers?  :)

I wonder if it's possible to script this in a B2BUA scenario?  I'm not sure
how one would do detection of 180 without SDP versus 180/183 with SDP in
B2B-land.  Or, what to do from there once it knew.


- Jeff



On Mon, Sep 23, 2013 at 10:43 AM, Jeff Pyle <jpyle at fidelityvoice.com> wrote:

> Hi Laszlo,
>
> Unfortunately the effect for the caller would be the same - ringback would
> stop.
>
> Here's the whole flow.  My terminating gateway is SIP to ISDN PRI.  Call
> terminates through the gateway to a particular mobile switching office.  I
> receive an ISDN PROGRESS message with inband audio.  This translates to the
> 183 with SDP.  Then I receive an ALERTING message with no inband audio.
>  This translates to the 180.  When the MSO sends the ALERTING, it has
> stopped sending the inband audio from the previous PROGRESS message.
>
> I'm thinking I need to do something else in the onreply_route to connect
> to the media server for a new 183.  Since I've executed t_relay to route
> the INVITE to the gateway, it seems my options are limited.
>
>
> - Jeff
>
>
>
> --
> Jeff Pyle <jpyle at fidelityvoice.com>
> Director, Voice Engineering
> Fidelity Voice and Data
> 216-245-4106
> www.fidelityvoice.com
>
>
>
> On Mon, Sep 23, 2013 at 8:57 AM, Laszlo <laszlo at voipfreak.net> wrote:
>
>> What if you simply drop the 180 in the onreply_route?
>>
>> -Laszlo
>>
>>
>> 2013/9/23 Jeff Pyle <jpyle at fidelityvoice.com>
>>
>>> Hello,
>>>
>>> I have one particular PSTN call flow that causes a 183 with SDP, then a
>>> 180 without SDP prior to 200 OK.  Some of my customer endpoints don't
>>> handle the 180 properly after a 183 and they cease to hear ringback.
>>>
>>> I'm thinking through how intercept the 180 and convert it to a 183 with
>>> SDP.  I have a media server available to generate the 183 and the media.
>>>  I'm struggling with how to relay the INVITE to the media server when the
>>> 180 arrives in the middle of the call setup.
>>>
>>> Any recommendations are appreciated.
>>>
>>>
>>>
>>> Regards,
>>> Jeff
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>> --
>>
>> --
>> Kind regards,
>> Laszlo Bekesi
>> http://voipfreak.net
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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