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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>
<br>
<blockquote
cite="mid:BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com"
type="cite">Btw, this is the latest SIP spec, better than the
draft spec for 183 you posted before.<br>
<br>
<a moz-do-not-send="true"
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
moz-do-not-send="true" href="mailto:boris@tagnet.ru">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 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
moz-do-not-send="true"
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 moz-do-not-send="true"
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
ЗАО "Тагнет"
тел. +7 (3435) 230001
факс +7 (3435) 230005
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Regards,
Boris
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