search for: ip_nat_sip

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=532 Summary: ip_nat_sip rewrote Call-ID instead of Contact - patch attached Product: netfilter/iptables Version: linux-2.6.x Platform: All URL: http://ibp.de/ OS/Version: All Status: NEW Severity: normal Priority: P2...

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=503 siqhamo@newlunar.co.za changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |ASSIGNED -- Configure bugmail: https://bugzilla.netfilter.org/bugzilla/userprefs.cgi?tab=email ------- You are

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=532 kaber@trash.net changed: What |Removed |Added ---------------------------------------------------------------------------- AssignedTo|laforge@netfilter.org |kaber@trash.net ------- Additional Comments From kaber@trash.net 2007-01-26 19:45 MET ------- (In reply to comment #0) >

I have four-interface Shorewall config set up. The "dmz" interface is bridged with "net" so I can assign public IP''s to the servers in the DMZ. I opted to do this rather than SNAT or ARP proxying because one of the servers runs Asterisk and SIP and NAT don''t always work well together. Somehow, my firewall config is causing a one-way audio problem in

I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for normal traffic, but I was unable to get the SIP phones to work. Using ngrep it was plain to see

Hello, Let me first start by saying Shorewall is awesome, and I use it everywhere from single box firewall, to home network firewall, even to our corporate firewall. I am experiencing a problem getting my home firewall to work with my BroadVoice VoIP connection. I use the Sipura SPA-2100 ATA (Analog Telephone Adapter) that came with my BroadVoice account. This happened when I tried to replace

Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to "reinvite" has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop

Dear Group! I want to improve the firewall rules for SIP and I already compiled the linux kernel with additional SIP netfilter settings Now I found this on the internet: modprobe ip_conntrack_sip ip_nat_sip Set IPtables filter rules iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A INPUT -p udp --dport 5060 -j ACCEPT Set IPtables NAT rules iptables -A FORWARD -o eth0 -p udp --dport 5060 -j ACCEPT iptables -t nat -A POSTROUTING -o eth0 -j SNAT --to-source ip.add.dr.ess -...

...l. I see no rejected packets hitting the firewall logs. I'm really at a loss as to what could be causing the calls to drop out for one party so regularly. Any clues where I could look further to debug this would be most useful. local firewall: modprobe ip_conntrack_sip ports=5060 modprobe ip_nat_sip # probably not needed since everything is forwarded: $IPTABLES -A FORWARD -s $INTERNAL_NET -d $ANYWHERE -p udp --dport 5060 -j accept-log # sip remote Asterisk server: $MODPROBE ip_conntrack $MODPROBE ip_conntrack_sip ports=5060 $IPTABLES -A INPUT -s $ANYWHERE -d $PUBLIC_ADDR -p udp --dport 5060...

Hi Everyone, I have a provider who is asking me to send SIP signals through 111.111.111.111 and then media through Media 1: 222.222.22.222 and Media 2: 244.244.244.244. This provider authenticates by IP and I think is using Sonus gear and hence they have some load balancer or something... I have always simply done this to work it out: host=111.111.111.111 peer=type and everything worked. But

...netbios_ns.o LD [M] net/ipv4/netfilter/ip_nat_h323.o LD [M] net/ipv4/netfilter/ip_nat_pptp.o CC [M] net/ipv4/netfilter/ip_nat_amanda.o CC [M] net/ipv4/netfilter/ip_nat_tftp.o CC [M] net/ipv4/netfilter/ip_nat_ftp.o CC [M] net/ipv4/netfilter/ip_nat_irc.o CC [M] net/ipv4/netfilter/ip_nat_sip.o CC [M] net/ipv4/netfilter/ip_tables.o CC [M] net/ipv4/netfilter/iptable_filter.o CC [M] net/ipv4/netfilter/iptable_mangle.o LD [M] net/ipv4/netfilter/iptable_nat.o CC [M] net/ipv4/netfilter/iptable_raw.o CC [M] net/ipv4/netfilter/ipt_hashlimit.o CC [M] net/ipv4/netfilter/ipt_...

Sorry but what does the ACL mean and its relation to the bindaddr? Regards Bilal > > 30 jan 2009 kl. 16.59 skrev Mike: > > > hI, > > > > Trying to understand how to setup two PRIs in > sip.conf. Using > > Asterisk 1.4.23. > > > > I have a provider giving me two PRI (different rate > centers) through > > SIP. Both PRI comes in from

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

Hello all, I have a linux machine with a SIP server (Asterisk) and 2 WAN interfaces (NATed) configured to do load balancing. I experienced problems with the SIP/RTP protocols and load balancing, because when initiating a call to an external SIP Host, a new RTP flow starts from the server to the Host, that sometimes uses another default route (due to the nexthop configuration). As i have two