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.<br><br>Thanks<br><br><br><div class="gmail_quote">On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis <span dir="ltr"><<a href="mailto:rob@hillis.dyndns.org">rob@hillis.dyndns.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">Emmanuel Pascal Bruno wrote:<br>
> I have a DID from IPKall.com which is forwarded to my asterisk box.<br>
> Then this extension should call my ip phone using Dial application.<br>
> Everything works fine, except when I pickup the phone, I can talk, the<br>
> other party can hear me, but I cannot hear anything the person says on<br>
> the ip phone.<br>
> Then after a couple of seconds, the call hangs up. I don't know why.<br>
><br>
> Here is the message I get:<br>
><br>
> SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918<br>
> -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10<br>
> [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum<br>
> retries exceeded on transmission<br>
> 4056be591b329cc9441f75b4560c3ccb@XX.XX.XXX.XX for seqno 102 (Critical<br>
> Response) -- See doc/sip-retransmit.txt.<br>
> [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging<br>
> up call 4056be591b329cc9441f75b4560c3ccb@XX.XX.XXX.XX - no reply to<br>
> our critical packet (see doc/sip-retransmit.txt).<br>
> == Spawn extension (ipkall, ipphone, 1) exited non-zero on<br>
> 'SIP/XX.XX.XXX.XX-09400918'<br>
><br>
> I am running asterisk 1.6 on CentOS<br>
><br>
> Please help me fix this<br>
<br>
</div></div>You likely have firewall issues since it appears that you are not<br>
receiving a response from the other end. Make sure you have *both* your<br>
SIP and RTP ports forwarded to your Asterisk box.<br>
<br>
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