Hello! 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?
Btw, this is the latest SIP spec, better than the draft spec for 183 you posted before.
http://www.ietf.org/rfc/rfc3261.txt
-Steve
On 30 June 2011 09:55, Boris Kovalenko <boris@tagnet.ru <mailto:boris@tagnet.ru>> wrote:
Hello!
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 :)
Is that all the signalling?
100 Trying only acknowledges the INVITE.
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.
The voice will only start when it sends 200/SDP which is when the call is answered.
-Steve
On 30 June 2011 06:00, Boris Kovalenko <boris@tagnet.ru <mailto:boris@tagnet.ru>> wrote:
Hello!
I have a customers with Cisco 2821 which want to use IVR on it. He has configured Cisco and told me there is no voice when number is called. The configuration is:
Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
I looked at tcpdump and the call flow is: FS (INVITE) --> Cisco 2821 Cisco 2821 (100 Trying) --> FS Cisco 2821 (RTP) --> FS
I think the call flow is wrong in that case and SIP100 can not go throught Freeswitch (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there should be SIP180 to get it work. But customers said this is OK and he has no troubles with same configuration with other provider. What may I do in this sutiation? Any specific Freeswitch configuration?
P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
-- Regards, Boris
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