[asterisk-dev] Question regarding progressinband
Kevin P. Fleming
kpfleming at digium.com
Mon Jun 27 12:56:08 CDT 2011
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
More information about the asterisk-dev
mailing list