[asterisk-dev] Question regarding progressinband

[Anatoliy Kounitskiy]

My bad :)

Asterisk 1.4.21.1:

Device                    Asterisk                     Device
-----------INVITE SDP--------->  |
<---------100 Trying------------     |
                                           | -----------INVITE SDP--------->
                                           | <---------100 Trying------------
                                           | <---------180 Ringing----------
<-----183 Session Prgoress--  |


Asterisk 1.4.2X+

Device                    Asterisk                     Device
-----------INVITE SDP--------->  |
<---------100 Trying------------     |
                                           | -----------INVITE SDP--------->
                                           | <---------100 Trying------------
                                           | <---------180 Ringing----------
<---------180 Ringing----------     |
<-----183 Session Prgoress--  |



On Mon, Jun 27, 2011 at 8:56 PM, Kevin P. Fleming <kpfleming at digium.com> wrote:
> On 06/27/2011 12:45 PM, Anatoliy Kounitskiy wrote:
>>
>> Hello,
>> I have question regarding the changes that are made in the sip
>> protocol in Asterisk - the option progressinband.
>> When this option is set to yes in asterisk version 1.4.21.1 - the call
>> flow is:
>>
>> sip.conf:
>> progressinband=yes
>>
>> Device                    Asterisk
>> -----------INVITE SDP--------->
>> <---------100 Trying------------
>> <-----183 Session Prgoress--
>>
>> After version 1.4.2X+ (tested with 1.4.36/1.4.41.1) the call flow changes
>> to:
>>
>> Device                    Asterisk
>> -----------INVITE SDP--------->
>> <---------100 Trying------------
>> <---------180 Ringing----------
>> <-----183 Session Prgoress--
>>
>>  From the information that I was able to found in the last version it
>> will always send 180 Ringing and after that the 183 Session Progress.
>
> You are only showing the incoming channel, not the outgoing channel or
> dialplan application(s) that are being used to handle this call. Without
> knowing what was happening in Asterisk, there is no way to guess why this
> sequence of messages was generated.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
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-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540