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">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">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>
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></div><div class="h5"><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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<br>
<br>
</div></div><pre cols="72">--
С уважением,
Борис Коваленко
ЗАО "Тагнет"
тел. +7 (3435) 230001
факс +7 (3435) 230005
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