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 <[EMAIL PROTECTED]> 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 > > [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 >
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