[Asterisk-Users] Grandstream Handytone 286 RTP Problems

[Mike Machado]

I guess that was another thing that was strange. When I talked, I saw no
RTP coming from the handytone to *. Would there be a reason the
handytone would not send RTP until it successfully received a RTP packet
from *, but since its not accepting RTP, it would not send it either? 

I do not even get one way communication, I get no way communication.


Does anyone out there have this firmware version of Handytone working at
all with *?

On Sun, 2004-01-04 at 20:55, John Baker wrote:
> I had a similar problem with a Cisco phone, i.e., the "Maximum retries
> exceeded on call" error.
> It took three days to track down the error to buggy network hardware.
> 
> Same symptoms, too - phone registered, one way conversation was ok (had a
> test extension
> for music on hold)
> 
> Fixed the hardware, phone works great.
> 
> John
> 
> ----- Original Message ----- 
> From: "Mike Machado" <mike at homelandtel.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Sunday, January 04, 2004 10:16 PM
> Subject: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
> 
> 
> > 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).
> >
> > sip.conf entry:
> >
> > disallow=all                    ; Disallow all codecs
> > allow=ilbc
> > allow=ulaw                      ; Allow codecs in order of preference
> >
> > [131]
> > type=friend
> > host=dynamic
> > reinvite=no
> > canreinvite=no
> > qualify=300
> > callerid="handytone <131>"
> > mailbox=131
> > nat=0
> >
> >
> > Handytone info:
> >
> > Software Version:    Program--1.0.4.17    Bootloader--1.0.0.11
> > HTML--1.0.0.19
> >
> >
> > Both on same subnet, no NAT. I have two Handytones, both exhibit same
> > symptoms.
> >
> > Anyone else have this problem?
> >
> >
> > -- 
> >
> >
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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