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

[Muhammad Shahzad Shafi]

Well, you have to sacrifice 183 Early Media, since converting 183 to
180 is far more easy and convenient then converting 180 to 183 (since
then you have to involve a media server, which is not going to be so
easy). 

Therefore, my advice would be to change all 183 from that
carrier to 180 response. You can use change_reply_status method,


http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status
[11] 

Thank you. 

On 2013-09-25 03:05, Jeff Pyle wrote: 

> 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 [10]> 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 [7]>
>> Director, Voice Engineering
>> Fidelity
Voice and Data
>> 216-245-4106
>> www.fidelityvoice.com [8]
>> 
>> On
Mon, Sep 23, 2013 at 8:57 AM, Laszlo <laszlo at voipfreak.net [9]>
wrote:
>> 
>>> What if you simply drop the 180 in the onreply_route?
>>>

>>> -Laszlo 
>>> 
>>> 2013/9/23 Jeff Pyle <jpyle at fidelityvoice.com
[3]>
>>> 
>>>> 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 [1]
>>>>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users [2]
>>> 
>>> --

>>> 
>>> -- Kind regards, Laszlo Bekesi http://voipfreak.net [4] 
>>>
_______________________________________________
>>> Users mailing
list
>>> Users at lists.opensips.org [5]
>>>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users [6]

-- 
Mit
freundlichen Grüßen
Muhammad
Shahzad
-----------------------------------
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email:
shaheryarkh at googlemail.com
 

Links:
------
[1]
mailto:Users at lists.opensips.org
[2]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[3]
mailto:jpyle at fidelityvoice.com
[4] http://voipfreak.net
[5]
mailto:Users at lists.opensips.org
[6]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[7]
mailto:jpyle at fidelityvoice.com
[8] http://www.fidelityvoice.com
[9]
mailto:laszlo at voipfreak.net
[10] mailto:jpyle at fidelityvoice.com
[11]
http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status
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