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 other party can hear me, but I cannot hear anything the person says on 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 retries exceeded on transmission <a href="mailto:4056be591b329cc9441f75b4560c3ccb@66.54.140.46" target="_blank">4056be591b329cc9441f75b4560c3ccb@XX.XX.XXX.XX</a> for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.<br>
[Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up call <a href="mailto:4056be591b329cc9441f75b4560c3ccb@66.54.140.46" target="_blank">4056be591b329cc9441f75b4560c3ccb@XX.XX.XXX.XX</a> - no reply to our critical packet (see doc/sip-retransmit.txt).<br>
== Spawn extension (ipkall, ipphone, 1) exited non-zero on 'SIP/XX.XX.XXX.XX-09400918'<br><br>I am running asterisk 1.6 on CentOS<br><br>Please help me fix this