Hi all, I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the phone is ringing, but when I pickup the call, there's no audio on both sides. I debugged the rtp-traffic at home. As long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its private address (192.168.100.10). After the softphone received this package, it tries to send RTP data to this address! Obviously those packages never reach asterisk... Does 'externip' just works for SIP and not for RTP? Where does the the internal IP-address come from and how can I set the right one? My configuration: [general] externip = 85.XXX.XXX.XXX nat = yes localnet = 192.168.100.0/24 [42] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=XXX qualify=yes port=5060 pickupgrouppermit=0.0.0.0/0.0.0.0 nat=yes mailbox=42 at device host=dynamic dtmfmode=rfc2833 dial=SIP/42 context=from-internal canreinvite=no callgroupcallerid=device <42> allow=alaw accountcodecall-limit=50 Regards Holger
I have no problem doing that by adding the information you have under [general] to /etc/asterisk/sip_nat.conf On Thu, Jan 29, 2009 at 7:30 AM, Holger Latz <tech at globalview.de> wrote:> > Hi all, > > I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone > in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the > phone is ringing, but when I pickup the call, there's no audio on both > sides. > > I debugged the rtp-traffic at home. As long as the phone is ringing, > everything is fine. But after the pickup, asterisk sends a SIP/SDP > package with its private address (192.168.100.10). After the softphone > received this package, it tries to send RTP data to this address! > Obviously those packages never reach asterisk... > > Does 'externip' just works for SIP and not for RTP? > Where does the the internal IP-address come from and how can I set the > right one? > > > My configuration: > > [general] > externip = 85.XXX.XXX.XXX > nat = yes > localnet = 192.168.100.0/24 > > [42] > deny=0.0.0.0/0.0.0.0 > disallow=all > type=friend > secret=XXX > qualify=yes > port=5060<http://0.0.0.0/0.0.0.0disallow=alltype=friendsecret=XXXqualify=yesport=5060> > pickupgroup> permit=0.0.0.0/0.0.0.0 > nat=yes > mailbox=42 at device > host=dynamic > dtmfmode=rfc2833 > dial=SIP/42 > context=from-internal > canreinvite=no<http://0.0.0.0/0.0.0.0nat=yesmailbox=42 at devicehost=dynamicdtmfmode=rfc2833dial=SIP/42context=from-internalcanreinvite=no> > callgroup> callerid=device <42> > allow=alaw > accountcode> call-limit=50 > > > Regards > Holger > > > > _______________________________________________ > -- 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/20090129/435c7dc2/attachment.htm
Ok, I found the problem. I suggested that I disabled completely my shorewall-firewall, because there were no rules loaded. But I were mistaken... shorewall loads some kernel-modules, especially ip_nat_sip and ip_conntrack_sip, and these modules interfere with asterisk! http://www.mail-archive.com/shorewall-users at lists.sourceforge.net/msg03968.html Regards Holger Holger Latz schrieb:> Hi all, > > I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone > in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the > phone is ringing, but when I pickup the call, there's no audio on both > sides. > > I debugged the rtp-traffic at home. As long as the phone is ringing, > everything is fine. But after the pickup, asterisk sends a SIP/SDP > package with its private address (192.168.100.10). After the softphone > received this package, it tries to send RTP data to this address! > Obviously those packages never reach asterisk... > > Does 'externip' just works for SIP and not for RTP? > Where does the the internal IP-address come from and how can I set the > right one? > > > My configuration: > > [general] > externip = 85.XXX.XXX.XXX > nat = yes > localnet = 192.168.100.0/24 > > [42] > deny=0.0.0.0/0.0.0.0 > disallow=all > type=friend > secret=XXX > qualify=yes > port=5060 > pickupgroup> permit=0.0.0.0/0.0.0.0 > nat=yes > mailbox=42 at device > host=dynamic > dtmfmode=rfc2833 > dial=SIP/42 > context=from-internal > canreinvite=no > callgroup> callerid=device <42> > allow=alaw > accountcode> call-limit=50 > > > Regards > Holger > > > > _______________________________________________ > -- 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