[asterisk-users] Need some help/Suggestions for multiple invites from Asterisk

Sat Sep 5 02:06:39 CDT 2009

Thank you for your response,
But we do get response from client (Step 2,3,4), the call is good, audio
DTMF everything works, except CDR is wrong; always 30-60 seconds more for
each call.

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,

On Fri, Sep 4, 2009 at 11:55 PM, Olle E. Johansson <oej at edvina.net> wrote:

>
> 5 sep 2009 kl. 04.58 skrev Jai Rangi:
>
> > 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.
>
> There's no response from the client phone.
> No 100 trying, no 180 ringing or 200 OK.
> We have to retransmit a few times and then just give up.
>
> Your client needs to wake up and start responding.
>
> Since the client was not responding, there never was a call to the
> client and no need to send a BYE.
>
> /O
>
>
> ---
> oej at edvina.net - http://edvina.net
> Open Unified Communication - building platforms with SIP and XMPP
>  From PBX to large scale implementations for carriers. Contact us today!
>
>
>
>
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