[asterisk-users] SIP registration fails

jonas kellens jonas.kellens at telenet.be
Thu Jun 25 14:49:33 CDT 2009


SIP-registration errors are solved by restarting the Asterisk-server.
But I expect them to return in time...  

I can make call now, but the other end does not hear me. So problem with
RTP-flow...

Can someone guide me to the solution ?

On the Asterisk-server I have this (iptables):

-A RH-Firewall-1-INPUT -p udp --dport 4569 -j ACCEPT
-A RH-Firewall-1-INPUT -p tcp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp --dport 11000:11500 -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 25 -j
ACCEPT
-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j
ACCEPT
-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited

In rtp.conf I have this :

rtpstart=11000
rtpend=11500

Asterisk is behind firewall. Endian firewall has following
configuration :

enable SIP proxy transparant
RTP port low : 11000
RTP port high : 11500

Firewall port forwarding : uplink:5060 >>> asterisk_ip:5060

Asterisk himself says :

    -- Executing [050510484 at intern:1] NoOp("SIP/grandstream-09813b58",
"via 3StarsNet") in new stack
    -- Executing [050510484 at intern:2] Dial("SIP/grandstream-09813b58",
"SIP/3starsnet/050510484") in new stack
    -- Called 3starsnet/050510484
    -- SIP/3starsnet-0981bf08 is making progress passing it to
SIP/grandstream-09813b58
    -- SIP/3starsnet-0981bf08 answered SIP/grandstream-09813b58
  == Spawn extension (intern, 050510484, 2) exited non-zero on
'SIP/grandstream-09813b58'

What do I need in sip.conf to overcome these rtp-problems ??
I have :
externip=78.21.62.99
canreinvite=no
jbenable = yes

[3starsnet]
type=peer
...
nat=yes
...


Thanks for the help !

Jonas.


On Thu, 2009-06-25 at 17:25 +0200, jonas kellens wrote:

> Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports
> opened and 5060 forwarded to Asterisk (192.168.2.2)
> 
> Can someone see why SIP-registration fails ??
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