[Asterisk-Users] Re: SIP and ATA186

[Mark Spencer]

Should be fixed in latest CVS.  If we send 183, we no longer send 180.

Mark

On Tue, 8 Apr 2003, Jim Gottlieb wrote:

> 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
>
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