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 > [EMAIL PROTECTED] 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 [EMAIL PROTECTED] - 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. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users