[asterisk-users] Call problems

[Emmanuel Pascal Bruno]

Oh ok, I knew it was something like that.  I have tried many different
settings on my router.  I'll dig into it some more.

Thanks


On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis <rob at hillis.dyndns.org> wrote:

> Emmanuel Pascal Bruno wrote:
> > I have a DID from IPKall.com which is forwarded to my asterisk box.
> > Then this extension should call my ip phone using Dial application.
> > Everything works fine, except when I pickup the phone, I can talk, the
> > other party can hear me, but I cannot hear anything the person says on
> > the ip phone.
> > Then after a couple of seconds, the call hangs up.  I don't know why.
> >
> > Here is the message I get:
> >
> >  SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918
> >     -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10
> > [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum
> > retries exceeded on transmission
> > 4056be591b329cc9441f75b4560c3ccb at XX.XX.XXX.XX for seqno 102 (Critical
> > Response) -- See doc/sip-retransmit.txt.
> > [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging
> > up call 4056be591b329cc9441f75b4560c3ccb at XX.XX.XXX.XX - no reply to
> > our critical packet (see doc/sip-retransmit.txt).
> >   == Spawn extension (ipkall, ipphone, 1) exited non-zero on
> > 'SIP/XX.XX.XXX.XX-09400918'
> >
> > I am running asterisk 1.6 on CentOS
> >
> > Please help me fix this
>
> You likely have firewall issues since it appears that you are not
> receiving a response from the other end.  Make sure you have *both* your
> SIP and RTP ports forwarded to your Asterisk box.
>
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