I have turned off firewall on the linux box, I have turned off firewall on the router I still have the same problem :-(
On Sat, Nov 1, 2008 at 1:22 PM, Emmanuel Pascal Bruno <[EMAIL PROTECTED]>wrote: > 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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