[Asterisk-Users] Grandstream Handytone 286 RTP Problems

[TeleSIP]

Hi Matteo,

I see a problem here.  And its the same as the trace I examined from
Wipeout.

After * sends back the "183 Session Progress" message, it should also send
the "STATUS 200 OK" once the call is answered (the only STATUS 200 OK I see
is the response to an INFO Message).  Since the GS never receives this
"STATUS 200 OK", it never sends back the ACK and the call will have a
"choppy" sound.

On the other file (cisco to GS), the "STATUS 200 OK" is clearly there and
shortly after you can see the ACK

Try to find out where the STATUS 200 OK is being lost.

Look at page #5 of the attached PDF to see that the STATUS 200 OK message is
indeed needed

Regards,
Andres.
http://www.telesip.net

----- Original Message ----- 
From: "Matteo Brancaleoni" <mbrancaleoni at espia.it>
To: <ricvil at telesip.net>
Sent: Wednesday, January 07, 2004 8:57 AM
Subject: Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems


> Hi.
> Here's the ethereal dumps (running on the * server):
> one is when calling from the GS to a cisco phone,
> the other is viceversa.
>
> I dumped only the traffic between the * server and the GS
>
> Running firmware 1.4.30
>
> Matteo.
>
> Il mer, 2004-01-07 alle 14:44, TeleSIP ha scritto:
> > Hi Matteo,
> >
> > Send me the Ethereal SIP Trace and I will take a stab at it.
> >
> > Regards,
> > Andres.
> >
> > ----- Original Message ----- 
> > From: "Matteo Brancaleoni" <mbrancaleoni at espia.it>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Wednesday, January 07, 2004 4:50 AM
> > Subject: Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
> >
> >
> > > Hi.
> > >
> > > I have the same issue with budgetones 102 (& 101) with firmware
1.0.4.30
> > > But happens also with .4.26 , .4.18 and .4.17 .
> > > Doing an ethereal trace, I noticed that the GS isn't answering to OK's
> > > sent by asterisk when the ringed party answers (GS doesn't not send
ACK
> > > to the cpnnection confirmation), so after x seconds (6, more or less)
> > > asterisk closes up the connection and so you get iCMP unreacheable
> > > on the RTP port (that's ok since * closed the port).
> > >
> > > So GS must fix that... send ACK to the 200/OK of the connection
> > > confirmation.
> > >
> > > Pretty interesting is that when you call to another * channel that's
not
> > > SIP (like ZAP,CAPI) all works ok... only SIP<->SIP raises the problem,
> > > or better only when the SIP call is initiated by the GS to another SIP
> > > device. If the call is started from a cisco to the GS, all works ok.
> > >
> > > Matteo.
> > >
> > > Il lun, 2004-01-05 alle 05:16, Mike Machado ha scritto:
> > > > I am trying to get the handytone 286 to make a very simple call to *
and
> > > > having problems. It registers with * just fine, but when I place a
call
> > > > (to echo test, for example), the RTP stream seems to have problems
> > > > opening. Here is there error I get in *:
> > > >
> > > > WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum
retries
> > > > exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26 at 192.168.2.6
for
> > > > seqno 0 (Response)
> > > >
> > > > When doing traces with ethereal, I see successful SIP and SDP
> > > > handshakes, but when * sends handytone RTP packets, I see a ICMP
Port
> > > > Unreachable messages sent from Handytone to * regarding the UDP RTP
> > > > packet. * then gives up and I see a BYE from *, which handytone
acks.
> > > >
> > > > Handytone config is default except obvious SIP registration
parameters.
> > > > I also have a Sipura SPA2000 and everything works perfect for that
one,
> > > > same extension and everything (not at same time of course).
> > > >
> > > >
> > > > Both on same subnet, no NAT. I have two Handytones, both exhibit
same
> > > > symptoms.
> > > >
> > > > Anyone else have this problem?
> > > -- 
> > > Matteo Brancaleoni
> > > Espia System Administrator
> > > Email : mbrancaleoni at espia.it
> > > Web   : http://www.espia.it
> > > Phone : +39 02 70633354      - ext 201
> > > IAX(2): guest at 213.140.14.155 - ext 201
> > > Iaxtel: 1-700-56-62458       - ext 201
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> Matteo Brancaleoni
> Espia System Administrator
> Email : mbrancaleoni at espia.it
> Web   : http://www.espia.it
> Phone : +39 02 70633354      - ext 201
> IAX(2): guest at 213.140.14.155 - ext 201
> Iaxtel: 1-700-56-62458       - ext 201
>
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