[Asterisk-Users] Grandstream Handytone 286 RTP Problems

TeleSIP ricvil at telesip.net
Wed Jan 7 06:44:03 MST 2004


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
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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