[asterisk-users] 180 Ringing missing

[Joshua C. Colp]

On Tue, Dec 1, 2020 at 11:24 AM marek <cervajs64 at gmail.com> wrote:

>
> Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a):
>
> On Tue, Dec 1, 2020 at 7:20 AM marek <cervajs64 at gmail.com> wrote:
>
>> hi,
>>
>> after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know,
>> its old. customer is very conservative...)
>>
>> i have problem with missing 180 Ringing
>>
>> flow is easy (PBX -> Asterisk -> SIP SBC)
>>
>> Asterisk 11
>> PBX - Asterisk
>> -> INVITE
>> <- 100 Trying
>> <- 183 Session Progress
>> ( <- RTP -> )
>> <- 180 Ringing
>> <- 200 OK
>>
>> Asterisk 13
>> -> INVITE
>> <- 100 Trying
>> <- 183 Session Progress
>> ( <- RTP -> )
>>
>> __MISSING RINGING___
>>
>> <- 200 OK
>>
>> temporarily i solved problem with using "R" param
>>
>> R: Default: Indicate ringing to the calling party, even if the called
>> party
>>      isn't actually ringing. Allow interruption of the ringback if early
>> media
>>      is received on the channel.
>>
>> it changed to
>>
>> Asterisk 13 (Dial(${ARG1},300,R)
>> -> INVITE
>> <- 100 Trying
>> <- 180 Ringing
>> <- 183 Session Progress
>> ( <- RTP -> )
>> <- 200 OK
>>
>> any ideas why Ringing is missing? any solutions?
>>
>
> Have you compared the signaling in both directions between the two
> versions to see if there is a difference?
>
> whats your goal with this question?
>
> asking if there are some side effects in incoming call ? (SBC -> Asterisk
> -> PBX)
>

 No side effects, but looking at the actual SIP signaling (sip set debug
on) and see what the remote side is sending for SIP responses as well.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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