Yes, either that or 200 (you can skip ringing to go straight to being answered).<br><br>Generally with 183/180 you'll get ringback but not hear voice in early media, you wouldn't normally hear voice until the 200 OK answers the call.<br>
<br>Media can't start up without SDP in the body of the 180/183/200. SDP contains the IP/port used for media and the selected codec. Media can't work without those details. Sometimes you'll see 180/183 without SDP there'll still be no early media containing ringback, in that case the client'll usually generate its own ringback but you'll always see SDP in the 200 OK.<br>
<br>-Steve<br><br><br><br><br><div class="gmail_quote">On 30 June 2011 13:00, Boris Kovalenko <span dir="ltr"><<a href="mailto:boris@tagnet.ru">boris@tagnet.ru</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<u></u>
<div text="#000000" bgcolor="#ffffff">
Hello!<br>
<br>
But even reading this RFC, I found that 100 Trying should be
followed by one of 18x response to get media channels connected,
isn't? <br><div><div></div><div class="h5">
<br>
<blockquote type="cite">Btw, this is the latest SIP spec, better than the
draft spec for 183 you posted before.<br>
<br>
<a href="http://www.ietf.org/rfc/rfc3261.txt" target="_blank">http://www.ietf.org/rfc/rfc3261.txt</a><br>
<br>
-Steve<br>
<br>
<br>
<br>
<div class="gmail_quote">
On 30 June 2011 09:55, Boris Kovalenko <span dir="ltr"><<a href="mailto:boris@tagnet.ru" target="_blank">boris@tagnet.ru</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
<div text="#000000" bgcolor="#ffffff"> Hello!<br>
<br>
Yes, this is all. INVITE, 100 and RTP. No other SIP
messages. So my knowledge of SIP if right. And customer lies
to me or forgot to configure something :)
<div>
<div><br>
<br>
<blockquote type="cite">Is that all the signalling?<br>
<br>
100 Trying only acknowledges the INVITE.<br>
<br>
That *may* be followed by 183 or 180 which'll indicate
ringing. The Cisco would only be able to send early
media if that was a 183/SDP or 180/SDP.<br>
<br>
The voice will only start when it sends 200/SDP which
is when the call is answered.<br>
<br>
-Steve<br>
<br>
<br>
<div class="gmail_quote">On 30 June 2011 06:00, Boris
Kovalenko <span dir="ltr"><<a href="mailto:boris@tagnet.ru" target="_blank">boris@tagnet.ru</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">Hello!<br>
<br>
I have a customers with Cisco 2821 which want
to use IVR on it. He<br>
has configured Cisco and told me there is no voice
when number is<br>
called. The configuration is:<br>
<br>
Cisco 2821 <-- FreeSwitch (full proxy mode)
--> PSTN<br>
<br>
I looked at tcpdump and the call flow is:<br>
FS (INVITE) --> Cisco 2821<br>
Cisco 2821 (100 Trying) --> FS<br>
Cisco 2821 (RTP) --> FS<br>
<br>
I think the call flow is wrong in that case and
SIP100 can not go<br>
throught Freeswitch<br>
(<a href="http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt" target="_blank">http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt</a>)
and there<br>
should be SIP180 to get it work. But customers
said this is OK and he<br>
has no troubles with same configuration with other
provider. What may I<br>
do in this sutiation? Any specific Freeswitch
configuration?<br>
<br>
P.S. FreeSWITCH Version 1.0.head (git-1c95ad9
2011-01-20 22-43-50 -0300)<br>
<br>
--<br>
Regards,<br>
Boris<br>
<br>
<br>
<br>
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<pre cols="72">--
С уважением,
Борис Коваленко
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