[Asterisk-Users] Re: SIP and ATA186

[Jim Gottlieb]

On 2003-04-08 at 09:42, Mark Spencer wrote:

> > So audio should probably be cut through as soon as dialing is finished.
> 
> We do pass audio through whenever we get 183 Session Progress (or even
> without it).  However, in at least one users tests, interfacing with Cisco
> equipment, the Cisco is sending to the wrong port number (not the one we
> specified in our SDP) so we're still very confused by that.

In my trace, it looks as though * is sending a 183 Session Progress,
immediately followed by a 180 Ringing.  This would explain why when
calling a busy number, I sometimes hear a slight blip of busy before I
hear ringing forever.

Is it possible * is sending 180 Ringing when it shouldn't?

Transmitting (no NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.40.90:5060
From: <sip:0054 at 198.51.175.9;user=phone>;tag=3862105745
To: <sip:18189950699 at 198.51.175.9;user=phone>;tag=as5a7814e5
Call-ID: 2798603792 at 192.168.40.90
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18189950699 at 198.51.175.9>
Content-Type: application/sdp
Content-Length: 191

v=0
o=root 14828 14828 IN IP4 198.51.175.9
s=session
c=IN IP4 198.51.175.9
t=0 0
m=audio 60794 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 192.168.40.90:5060
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.40.90:5060
From: <sip:0054 at 198.51.175.9;user=phone>;tag=3862105745
To: <sip:18189950699 at 198.51.175.9;user=phone>;tag=as5a7814e5
Call-ID: 2798603792 at 192.168.40.90
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18189950699 at 198.51.175.9>
Content-Length: 0