[asterisk-users] Asterisk answers inbound call during ringing

[Dovey Forman]

I am running Trixbox PRO.



I don’t know if this is a config issue, since it would seem to be odd that
an inbound SIP call into asterisk would answer the call even during ringing.



Check out the SIP trace below.



It’s a call from the PSTN into an asterisk DID assigned to an ext.



On the PSTN side the caller heard ringing before the call was answered…yet
on the SIP side there is no 18x response back just:



Here is a trace of the INBOUND side (PSTN into SBC) and OUTBOUND (SBC into
Asterisk):



INCOMING INTO SBC



Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y]  INVITE
sip:2015555555 at y.y.y.ySIP/2.0

                                                                m=audio
20828 RTP/AVP 0 18 8 101 c=IN IP4 x.x.x.x

                                                                a=fmtp:18
annexb=no

Feb 17 15:01:40 2010 [y.y.y.y] ==> [x.x.x.x]  SIP/2.0 100 Trying

Feb 17 15:01:40 2010 [y.y.y..y] ==> [x.x.x.x]  SIP/2.0 200 OK

                                                                m=audio
21642 RTP/AVP 0 c=IN IP4 y.y.y.y

Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y]  ACK sip:y.y.y.y:5060 SIP/2.0

Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y]  ACK sip:y.y.y.y:5060 SIP/2.0

Feb 17 15:02:31 2010 [y.y.y.y] ==> [x.x.x.x]  BYE sip:2015555555 at x.x.x.x:5060
SIP/2.0

Feb 17 15:02:31 2010 [x.x.x.x] ==> [y.y.y.y]  SIP/2.0 200 OK



OUTGOING FROM SBC INTO ASTERISK



Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y]  INVITE
sip:2015555555 at y.y.y.y:5060;user=phone
SIP/2.0

                                                                m=audio
21142 RTP/AVP 0 18 8 101 c=IN IP4 x.x.x.x

                                                                a=fmtp:18
annexb=no

                                                                a=sendrecv

Feb 17 15:01:40 2010 [y.y.y.y] ==> [x.x.x.x]  SIP/2.0 100 Trying

*Feb 17 15:01:40 2010 [y.y.y.y] ==> [x.x.x.x]  SIP/2.0 200 OK*

*                                                                m=audio
19282 RTP/AVP 0 8 c=IN IP4 y.y.y.y *

Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y]  ACK sip:2015555555 at y.y.y.ySIP/2.0

Feb 17 15:02:31 2010 [y.y.y.y] ==> [x.x.x.x]  BYE sip:2015555555 at x.x.x.x:5060
SIP/2.0

Feb 17 15:02:31 2010 [x.x.x.x] ==> [y.y.y.y]  SIP/2.0 200 OK
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100217/a2ad22fe/attachment.htm