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List:       asterisk-users
Subject:    Re: [asterisk-users] Turn off SIP 183 Session Progress in Asterisk
From:       "Andrew Joakimsen" <joakimsen () gmail ! com>
Date:       2007-07-31 23:36:53
Message-ID: 23fd749a0707311636q70af1556p7dfb96906bf06f07 () mail ! gmail ! com
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On 7/31/07, Richard Brady <rnbrady@gmail.com> wrote:
>
> [Resent due to non-descriptive subject line.]
>
> Hi folks
>
> When connecting two SIP users, is there any way to stop Asterisk from
> sending SIP 183 Session Progress messages, either globally or
> per-peer?


Yes, the option is progressinband in sip.conf, see:
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband

Scenario as follows:
>   Call from UA1 to Asterisk (UA2) to UA3.
>   UA3 sends RTP before SIP OK to Asterisk (UA2).
>   Asterisk (UA2) detects early audio from UA3 and sends 183 Session
> Progress with SDP to UA1.
>
> Instead I would like it to just send on the early audio, is this possible?


 Why do you care about the 183 or not? Because ignoring anything you said
about SIP 183 what you want is to send SIP 183 which would indicate there is
inband indications. See:
http://www3.ietf.org/proceedings/99jul/slides/mmusic-sip183-99jul/index.htm

Also it wouldnt hurt to read the SIP RFC's to have a better understanding of
what is going on:

ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt
ftp://ftp.rfc-editor.org/in-notes/rfc2543.txt

[Attachment #5 (text/html)]

<br><br><div><span class="gmail_quote">On 7/31/07, <b \
class="gmail_sendername">Richard Brady</b> &lt;<a \
href="mailto:rnbrady@gmail.com">rnbrady@gmail.com</a>&gt; wrote:</span><blockquote \
class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt \
0pt 0.8ex; padding-left: 1ex;"> [Resent due to non-descriptive subject \
line.]<br><br>Hi folks<br><br>When connecting two SIP users, is there any way to stop \
Asterisk from<br>sending SIP 183 Session Progress messages, either globally \
or<br>per-peer?</blockquote> <div><br>Yes, the option is progressinband in sip.conf, \
see: <a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband \
">http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband</a><br> \
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, \
204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Scenario as \
follows:<br>&nbsp;&nbsp;Call from UA1 to Asterisk (UA2) to UA3.<br>&nbsp;&nbsp;UA3 \
sends RTP before SIP OK to Asterisk (UA2). <br>&nbsp;&nbsp;Asterisk (UA2) detects \
early audio from UA3 and sends 183 Session<br>Progress with SDP to \
UA1.<br><br>Instead I would like it to just send on the early audio, is this \
possible?</blockquote><div><br>&nbsp;Why do you care about the 183 or not? Because \
ignoring anything you said about SIP 183 what you want is to send SIP 183 which would \
indicate there is inband indications. See:  <a \
href="http://www3.ietf.org/proceedings/99jul/slides/mmusic-sip183-99jul/index.htm">htt \
p://www3.ietf.org/proceedings/99jul/slides/mmusic-sip183-99jul/index.htm</a><br><br>Also \
it wouldnt hurt to read the SIP RFC&#39;s to have a better understanding of what is \
going on: <br><br><a \
href="ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt">ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt</a><br><a \
href="ftp://ftp.rfc-editor.org/in-notes/rfc2543.txt">ftp://ftp.rfc-editor.org/in-notes/rfc2543.txt</a>
 <br><br></div><br></div><br>



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