similar to: netfilter conntrack mangling canreinvite?

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

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 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

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

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

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) >

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

http://bugzilla.netfilter.org/show_bug.cgi?id=757 Summary: SIP connection helper not setting RTCP conntrack expectation Product: netfilter/iptables Version: linux-2.6.x Platform: i386 OS/Version: Ubuntu Status: NEW Severity: normal Priority: P5 Component: ip_conntrack

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

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

I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to

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

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

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, 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

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

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

2009/2/5 Olivier <oza-4h07 at myamail.com> > Hi, > > Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a > table listing ATA/Gateways combinations. > Could anyone successfully set a Patton M-ATA to work with another one, > using Asterisk 1.4 ? > > Is reinvite (canreinvite=yes) necessary or not ? > > Regards > > Replying to myself, I

Hi, I can not get this working: Asterisk on public IP. Two SIP phones behind NAT - in the same LAN. I works perfectly (two way sound) when each peer (friend) can not reinvite - audio stream goes through Asterisk. The problem pops up when I define canreinvite=yes on each peer definision so I suppose to stream audio directly between phones (in the same local LAN). Right after called party

Howdy, Is it possible to send a reinvite after the media has connected? Scenario: Inbound call hits asterisk ivr then is sent out to an extension using the dial command. We have to carry the rtp streams in this case as asterisk cant send the reinvite after the ivr has stopped playing the message as we already connected the call. Question: Any way around this or is there a better way we can do