similar to: Calls drop after a couple of minutes.

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

=20 Andrew Fenn wrote: > You don't need their program to use justvoip, voipdiscount, etc=2E You > can use any sip client to connect to Betamax servers=2E Try Twinkle=2E >=20 > On Mon, Jul 27, 2009 at 11:24 PM, miroa84<wineforum-user at winehq=2Eorg> wrote: >=20 > > I tried to install justvoip several times and I cannot install it=2E Can somebody tell me how to

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

I tried to install justvoip several times and I cannot install it. Can somebody tell me how to install it on ubuntu? Meybe next version of WINE will support it?

I have noticed that Asterisk (version 1.2.13) is not hanging up a call when the wifi handset moves out of range. My setup is Nokia E61 connected to wifi access point (private IP range) and then to server on internet (public IP). I have been testing using the talking clock application, and walking out of range does not hang up the call. The call will continue for hours even though the handset

I am new user of WINE and need a little help. 1. From the best I can tell I have everything configured correctly. However I can't figure out how to get a Windows screen to show up. 2. How do I get a WINE icon over to my desktop so as start it with a mouse click(s) Yes as you can tell I am very new to linux systems as I just installed this system this afternoon, so please be genle with

I am the IT Development Specialist for a small community college and our CIO has asked me to explore an alternative to Microsoft Active Directory as we are separating from our parent university and funding is tight so we were looking into CentOS with 389 Directory Server. Any advise or suggestions would be very helpful. Jacob Tennant

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

From: "Guillermo Salas M." <gsalas at manta.telconet.net> > http://www.gizmo5.com/opensky Free calls are available up to 5 > minutes. If you need longer calls there's a commercial service you can > purchase. > Can be used to receive calls from skype? Yes it can. For example anyone who calls me now on Skype at michaelGizmo5 it will ring the IP phone connected to

Hi, I noticed recently when installing the GDD package for R under GNU/Linux that it required the gd library (http://libgd.org/) for generating graphics. The resolution of this was to simply install the library on my system, and then GDD successfully installed without any complaints. However, the variant of GNU/Linux that I use is Gentoo, so I filed a bug requesting that a USE flag be set for

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

On Wed, May 03, 2017 at 02:35:08PM +0800, Bright Zhao wrote: > The use case the shared default gateway for multi-tenant, if that the case the node who own the default gateway will have problem to route with different tenant who has overlapped address scope? Is it true when no any other tools like the namespaces? > > (tenant1)\ > (tenant2)——common node—— shared gw node—— Internet >

Adding back the nouveau list... On Sun, Feb 9, 2014 at 8:28 AM, Attila T?th <tothsoft at gmail.com> wrote: > Linux mint 16 (64bit) > > Repo: Ubuntu X-SWAT <ubuntu-x at lists.ubuntu.com><ubuntu-x at lists.ubuntu.com> > Mesa: 10.2.0~git20140205.44338cd8-0ubuntu0sarvatt~saucy > libdrm: 2.4.52+git20140121.46d451c9-0ubuntu0sarvatt~saucy > libg3dvl-mesa:

>Just to check, have you set up >srvlookup=yes > >under the general context in your sip.conf? > >Alyed No, but I put it now but the result is the same. And googleing further https://issues.asterisk.org/view.php?id=3723, it seems that is an old issue... Don't know for witch version is, 1.2?... But is what is happening to me. I'm putting bind in the asterisk server and

Just for the sake of this thread I'll paste part of the last post regarding this issue in the asterisk bug tracker. kpfleming on 2005-03-10 post: "Essentially, what we are saying is that if you are going to use DNS to resolve critical information in your Asterisk configuration, you need to do everything possible to ensure that the DNS lookups will not block for long periods of time.

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

Hi, Guus The use case the shared default gateway for multi-tenant, if that the case the node who own the default gateway will have problem to route with different tenant who has overlapped address scope? Is it true when no any other tools like the namespaces? (tenant1)\ (tenant2)——common node—— shared gw node—— Internet (tenant3)/ But if the each tenant have it’s dedicate default gateway, but

I am trying to get multicast traffic to traverse 3 different subnets call connected to the same linux router. This is primarily to get rendezvous/zeroconf services working for Macs on the network. Being able to experiment with the VideoLAN client''s multicasting abilities would be a bonus. I see that rendezvous sends out packets with a TTL of 1 and expects them back with a TTL of 255 so

hauptsturmfuehrer hitchcocks gursin 1nternet Pharmacy & 0vernight Shi,pping! They got X. at nax, and V.aliu.m, Phe-ntermine, S0ma..and other popular products.. It's private, fast, and safe. OverNiteDelivery! C Y http://gxusyu.bgm.sewerage5665pills.us/f74/ Cynthia told the group that she originally took a Circle K clerk job as an interim position while she looked for something

I'm looking for a recipe for a 3 way call where one of the parties can (without using the flash button) dial-out and add a third participant to the call. I tried Googling but it seems I'm missing a key search term. The reason I wanted to avoid using the flash button is that some handsets don't have it (nokia E61 who's 2 way calling via sip is also broken) Something like: 1.