similar to: SIP AND NAT

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

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

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

Hi all. I can''t seem to get the h323 connection tracking configured correctly for Shorewall. I am using the Debian Shorewall 4.5.16.1 package. I am running a Debian 3.9 kernel like so: # uname -a Linux gw 3.9-1-amd64 #1 SMP Debian 3.9.8-1 x86_64 GNU/Linux My version of iptables is: # iptables -V iptables v1.4.20 If I add the following rule in the /etc/shorewall/tcrules file to

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

hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this?

Dear Devs, I have x4 4TB HDDs formatted with: mkfs.btrfs -L bu-16TB_0 -d raid1 -m raid1 /dev/sd[cdef] /etc/fstab mounts with the options: noatime,noauto,space_cache,inode_cache All on kernel 3.8.13. Upon using rsync to copy some heavily hardlinked backups from ReiserFS, I''ve seen: The following "block rsv returned -28" is repeated 7 times until there is a call trace

Hi, I'm running CentOS 6.5 as NFS server (v3 and v4) and exporting Ext4 and XFS filesystem. After many months that all works fine today the server crash: Jan 30 09:46:13 qb-storage kernel: ------------[ cut here ]------------ Jan 30 09:46:13 qb-storage kernel: WARNING: at lib/list_debug.c:26 __list_add+0x6d/0xa0() (Not tainted) Jan 30 09:46:13 qb-storage kernel: Hardware name: PowerEdge

http://bugzilla.netfilter.org/show_bug.cgi?id=714 Summary: Kernel panics in same_src() Product: netfilter/iptables Version: linux-2.6.x Platform: x86_64 OS/Version: All Status: NEW Severity: normal Priority: P5 Component: NAT AssignedTo: netfilter-buglog at lists.netfilter.org ReportedBy:

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

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

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

Hello,   Are there general guidelines around on how to configure Shorewall for use with SIP phones ?  Especially regarding (some?) Cisco SIP phones which are expecting a reply at port 5060 while sending from an arbitrary high port. Thanks ! ------------------------------------------------------------------------------ Master Visual Studio, SharePoint, SQL, ASP.NET, C# 2012, HTML5, CSS, MVC,

Hi All, I am trying to compile the latest xen source code. But i am unable to compile it successfully. It seems there is some bug in /linux-2.6.18-xen.hg/net/ipv4/netfilter/ipt_ecn.c file. Can anybody suggest what do i need to do ? i tried with xen-3.3.0 as well as xen-3.3.1. If i wish to make a similar xen-enabled kernel which comes with fedora 8 or SUSE11, where should i place the

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

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

Hey, we're experiencing a weird problem with Asterisk 1.8.13.1 (1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via a PBX (sipgate.de) work perfectly fine, almost 100% of the time. However, calls that are routed to sipgate.de, which then routes the call back to our Asterisk instance are "silent" most of the time. What I mean with that is that even though RTP traffic