[asterisk-users] Call problems

Rob Hillis rob at hillis.dyndns.org
Sat Nov 1 13:04:47 CDT 2008

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 
> 4056be591b329cc9441f75b4560c3ccb at XX.XX.XXX.XX 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 4056be591b329cc9441f75b4560c3ccb at XX.XX.XXX.XX - 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.

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