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

Thu Sep 26 19:33:27 CEST 2013

Yes of course, you need to remove sdp as well while changing reply
from 183 to 180,

http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#id292832
[15] 

Thank you. 

On 2013-09-26 18:21, Jeff Pyle wrote: 

> Muhammad,

> 
> That makes sense. I think in my case I would have to strip the SDP
as well? Any thoughts on the media sent from the b-leg back to the a-leg
when it's not being expected (because there is no SDP)? 
> 
> - Jeff 
>

> On Tue, Sep 24, 2013 at 11:03 PM, Muhammad Shahzad Shafi
<shahzad at voip-demos.com [14]> wrote:
> 
>> 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
>> 
>>
_______________________________________________
>> Users mailing list
>>
Users at lists.opensips.org [12]
>>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users [13]

-- 
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:
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[1]
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[2]
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[3]
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[4] http://voipfreak.net
[5]
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[6]
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[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
[12]
mailto:Users at lists.opensips.org
[13]
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[14]
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[15]
http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#id292832
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