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 [050510...@intern:1] NoOp("SIP/grandstream-09813b58",
"via 3StarsNet") in new stack
-- Executing [050510...@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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