search for: ip_conntrack_sip

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

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

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

...nat'ed network couldn't complete outgoing calls. I would get initial audio, but the call was never connected as far as the softphone was concerned. Analysis showed that ip_nat_sip rewrote the IP-Address in the Call-ID: instead of the IP-Address in the Contact: header. The problem is in ip_conntrack_sip.c:skp_epaddr_len: it searches for the next @ to skip the username, but does not stop at the end of the header line. In my case, SJPhone sends a Contact without a username, and the next @ was in the Call-ID header. Attached is a (trivial) fix. The fix should be safe, even in the presence of cli...

...erisk 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 another day into verifying this, may I ask if anyone else has had a similar problem and found a solution? It appears conntrack is rewriting the SDP so that the address is reverted to the PBX address. Here are the relevant SDP portion of a reinvite captured on the PBX using tcpdump...

...everything on the NAT gateway/firewall. 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 $P...

Hi. there is a way to MARK udp VOIP (SIP) traffic, in order to put in a highest prio class ? Traffic flow seems start on udp 5060 port, but next both server and client seems jump to a random(?) port. I can''t use CONNMARK because is udp traffic. I only see a pattern for L7 patch in order to SIP traffic identification , but I run 2.4 kernel series . When you patch 2.4 kernel with

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 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

Linux does not have it's own sip/h323 modules (ip_conntrack_sip and ip_conntrack_h323), however I have found these modules available in the Linksys WRT54GS open source firmware. Would it be legal to use these modules with another Linux distribution (eg, RedHat, Gentoo, Debian..)? -- Chris Hills IT Services North East Worcestershire College

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.ad...

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 Guys, i wanted to share this with u and ask for little help at the same time: i used iptables to secure my server, so i wnet ahead and blocked avery thing except a couple of domain protocols and UDP ports of SIP, IAX2 and that range 15000 to 20000, tested it and OK. when in production, the calls were taking a huge time 7s to be established and somtimes after call setup people cannot hear ech

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

Hi All, Until now I've only used IAX2 to connect to ITSPs. I've been toying with a SIP connection to Gizmo Project, but not yet successfully. It brings to mind a question. At what point does it make sense to consider a SIP-aware firewall such as those from Ingate? I'd hate to move away from my m0n0wall, which is open source, easy to manage and has served me brilliantly for two

The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension number&password could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there