[asterisk-users] Call problems
Emmanuel Pascal Bruno
tipascal at gmail.com
Sun Nov 2 11:49:32 CST 2008
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 <tipascal at gmail.com>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 <rob at hillis.dyndns.org> 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
>> > 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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