[asterisk-users] Need some help/Suggestions for multiple invites from Asterisk
Jai Rangi
jprangi at gmail.com
Fri Sep 4 21:58:26 CDT 2009
Hello,
I have a issue between asterisk and windows based VoIP system (Client).
Vendor SIP Server --> My asterisk --> Client
Here is ethereal trace between asterisk and client.
1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525648 at 192.168.4.23 <sip%3A1978525648 at 192.168.4.23>, with session
description
2 0.042380 192.168.4.23 -> 192.168.3.222 SIP Status: 100 Trying
3 0.044235 192.168.4.23 -> 192.168.3.222 SIP Status: 183 Session
Progress
4 0.046546 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, with
session description
5 0.046752 192.168.3.222 -> 192.168.4.23 SIP Request: ACK
sip:1978525648 at 192.168.4.23:5060
So far so good, call is established and audio conversations starts.
But next my asterisk is sending Invite again and again and again,
6 0.047036 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525648 at 192.168.4.23:5060, with session description
7 0.266230 192.168.3.222 -> 192.168.4.23 RTP Payload type=ITU-T G.729,
SSRC=905761218, Seq=56540, Time=0
8 1.046087 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525648 at 192.168.4.23:5060, with session description
9 2.046091 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525648 at 192.168.4.23:5060, with session description
10 4.046102 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525648 at 192.168.4.23:5060, with session description
I disconnected the call, Receive BYe from Vendor, Asterisk acknowledge Bye
and does not send Bye to the client. Few more invites from Asterisk to the
client machine.
11 8.046123 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525648 at 192.168.4.23:5060, with session description
12 16.046179 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525648 at 192.168.4.23:5060, with session description
After a 30 second wait, asterisk receive Bye from Client.
13 24.253811 192.168.4.23 -> 192.168.3.222 SIP Request: BYE
sip:6056929587 at 192.168.3.222 <sip%3A6056929587 at 192.168.3.222>
14 24.253975 192.168.3.222 -> 192.168.4.23 SIP Status: 200 OK
15 32.046319 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525648 at 192.168.4.23:5060, with session description
16 32.085897 192.168.4.23 -> 192.168.3.222 SIP Status: 100 Trying
17 32.090654 192.168.4.23 -> 192.168.3.222 SIP Status: 183 Session
Progress
18 32.092666 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, with
session description
19 32.593335 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, with
session description
20 33.607552 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, with
session description
I am using canreinvite=yes, (Must use that to avoid media going through my
asterisk server.
I dont have any issue if asterisk send call to another asterisk box.
Can some one please shed some light why asterisk is sending multiple
invites.
Best,
-Jai
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090904/4c000d78/attachment.htm
More information about the asterisk-users
mailing list