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<4056be591b329cc9441f75b4560c3ccb at 66.54.140.46>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<4056be591b329cc9441f75b4560c3ccb at 66.54.140.46>-
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
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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
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I have tried that too with no results On Sun, Nov 2, 2008 at 1:30 PM, Rob Hillis <rob at hillis.dyndns.org> wrote:> Emmanuel Pascal Bruno wrote: > > I have turned off firewall on the linux box, I have turned off > > firewall on the router I still have the same problem :-( > > Disabling firewalls is almost certainly going to ensure the problem > persists. You need to ensure that all SIP and RTP ports are > port-forwarded from your firewall 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081102/a16a0d36/attachment.htm
Emmanuel Pascal Bruno wrote:> I have tried that too with no results > > > On Sun, Nov 2, 2008 at 1:30 PM, Rob Hillis <rob at hillis.dyndns.org > <mailto:rob at hillis.dyndns.org>> wrote: > > Emmanuel Pascal Bruno wrote: > > I have turned off firewall on the linux box, I have turned off > > firewall on the router I still have the same problem :-( > > Disabling firewalls is almost certainly going to ensure the problem > persists. You need to ensure that all SIP and RTP ports are > port-forwarded from your firewall to your Asterisk box. > > _______________________________________________Maybe you use wireshark to diagnose what is going on in your network? Kind regards Eberhard