similar to: SIP/H323 modules for netfilter

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

Hi I noticed in a post that Timo might start work on a LDA Great! If so, here is something I would really like included. Support for PAM There is quite a lengthy reason behind this. In my organisation, user accounts are created on ldap. It is up to the services to create home directories whenever the user first uses them. For instance, Samba will do this the first time they connect to their

dovecot-0.99.10.5-0.FC2 I have had a report from a user that when accessing their inbox, the connection is suddenly dropped. Dovecot reports the following:- imap(user): file buffer.c: line 357 (buffer_set_start_pos): assertion failed: (abs_pos <= I_MIN(buf->used, buf->limit)) From searching the list, I found an indication that it is a problem with an individual message. As the user

I have come across a strange situation, involving new accounts. I am running Dovecot 0.99.10-5 on Fedora Core 2. Dovecot is configured to use pam for authentication, and the dovecot pam service is configured to use pam_mkhomedir. When an account is created and the user logs in through SquirrelMail, there is no problem. The standard folders are available - Inbox, Drafts, Sent Items and Trash.

Hi, We have 2 offices interconnected with a VPN. This is the policy file in both of the Firewalls: fw loc ACCEPT loc fw ACCEPT #fw net DROP info fw net ACCEPT loc net DROP info loc vpn ACCEPT vpn loc

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

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, I have installed tinc-1.0pre7-1.i386.rpm in redhat 7.2, followed the documentation. While running daemon, its giving following error message Jul 1 15:06:18 localhost modprobe: modprobe: Can't locate module char-major-36 Jul 1 15:06:18 localhost tinc.percipia[1292]: Could not open /dev/misc/net/tun: Nos uch device ================== /etc/modules.conf alias char-major-10-200 tun

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

i apologize if this has been asked before, but things are too busy to preclude a full search of the list. i have both cable and dsl from the local providers here. due to wiring issues here, i''ve been forced to put the cable modem in one end of the house and the dsl modem in the other. the cable modem is firewalled off by a cisco pix 501 (192.168.2.12). the dsl modem is firewalled off

Hello, I have a very strange problem that I can''t seem to be able to figure out. First of all: my network scheme: http://heavyg.safehex.be/network.png I have an agreement with my neighbour that I can use his wireless network to experiment with extra internet uplinks in exchange for some filesharing etc. Now I have set up a wireless router at my place, that connects to his network.

Hi, what's the difference between h323 and oh323 modules? which one should I use? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030722/3d3edb73/attachment.htm

hi, there is no support for patch-o-matic netfilter modules. what i have to do if i want to use several patch-o-matic modules? which parts of code has to be changed and will that changed be included into the main shorewall tree in future or not? best regards claus

I have asterisk running in a wrt54gs attached is a pap2 with 2 extensions working on it, the problem now is that there is lots of echo, some rythm in the background, and the voice is delayed by about 4 or 5 sec's between the 2 extensions. memory usage is about 15 to 20 megs so I think I can solve the problem with correct settings, anyone know where I might start to correct these issues

Can anyone inform me if voip can be used on a IPv6 network? Does any hard phones/soft phones/Asterisk support it? Google told me that there was/is a bounty on it, but that expired august last year. Furthermore, there used to be a patch (Bernhard Schmidt), but that one is about a year old. I presume it can't be used on recent versions of "*" Hans -- pgp-id: 926EBB12

hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns "SIP/2.0 404 Not Found" any ideas ?

Hello folks: This may have been discussed many times before but I have not been able to find it. I have also not been able to resolve it myself so I am asking here with hope that someone can straighten me out. I am using ip to do multisource policy routing or two connections to the internet. I have a linksys wrt54gs route which connects two machines by wire through the switch and three

Hi, First at alll, I beg your pardon because maybe I explained bad my questions (because my low level english) I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2 compiled and installed. I have modules alsa 0.9.8 compiled and installed My PC has an audio card ac97 chipset intel i810 in its motherboard. I want to use asterisk in this way: |PC linux | --ethernet--

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

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