similar to: SIP extension outside of IP tables firewall

I don't have a CCO account with cisco and I want to use asterisk for my cisco 7960 phones But I could not get myself a firmware image for it. If anyone can share a copy of it, would be most appreciated Or if you could point me to where I can download it Thank you ,jm -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, new to asterisk and trying to set it up to work with my voip provider (vbuzzer.com). I am behind a firewall that I don't have access to, to open ports etc. Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun. However I did have to enter the outboundproxy server to get them to function. Not

Hi, I was woredering if you could help me to put into practice this solution. The idea: Create a IVR-Voicemail The scene: PSTN------/6------PBX--------/12--------- Internos | /4 ports | IVR-Voicemail The Operation: 1)Where a call enters from the PSTN, the PBX

I'm interested to hear how folks are handling NAT SIP routing issues "in the wild" for commercial use. Are you using a commerical SIP-aware NAT router solution? If so, what? Are you using a software SIP-proxy like SER or siproxd? If so, which? Do you set everything to "canreinvite=no" in sip.conf? Any comments about real-world implementations would be welcome. Thanks

I have three ethernet card and two of them connected to diffrent ISP and has diffrent gateways. And one ethernet card is connected to local lan. I have installed iptables and configured the firwall to work as nat gateway for the users. Now I want to route only mail traffic to the perticular ISP and which I am not able to do it. I have done iproute configuration what ever lartc said. But still I

Hi I install shorewall and every seed ok but when I start firwall I got the = followings .......... /sbin/shorewall: line 521: --: command not found .......... .......... Regards, John Fong

Hi I am running firewall/router "brew" made of RedHat, Shorewall, Siproxd and GnuGK on a box that connects through PPPoE to Internet. I run Asterisk on another box behind of it and it seem to work fine for me. I am thinking of replacing the router box because hardware is getting flaky. I do not want to go through pain of assembling all this stuff together again. Does anybody know of

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

Our Client software used samba protocol,If there is no firewall case , this protocol work properly from remote site.All of firwall configuration open ( this mean is all of port open ). Samba work very well, but if some constrain put in, it doesn't work. 1)I would like to know about port usage of samba protocol. I think snmpd, nmpd daemon used some port number. i can see /etc/service file.

Hello VoIP guys Sorry for being somewhat off-topic. At the moment I am studying informatics in the seventh semester and I need to start thinking about my thesis. As I am very interested in VoIP technologies I thought about picking this as my main topic. So far I have only little experience in this area. I have been fiddling around with siproxd and pfSense and have red the one or the other packet

Hello all, I am about to send off a manuscript and, although I am fairly confident I have used the lme function correctly, I want to be 100% sure. Could some kind soul out there put my mind at ease? I am simply interested in whether a predictor (SPI) is related to height. However, there are five different populations, and each may differ in mean level of height as well as the relationship

I'm running Asterisk on an external static IP address, siproxd on a different server with its own external static IP address, and communicating using a Grandstream behind a NAT firewall configured to register with Asterisk using siproxd as the outbound proxy. Now I'm aware that siproxd is not intended to be used as an outbound proxy but rather as a SIP relay when installed on the same box

http://bugzilla.netfilter.org/show_bug.cgi?id=577 Summary: cannot set spi/reqid numbers higher than 0x7fffffff (policy match) Product: iptables Version: unspecified Platform: i386 OS/Version: All Status: NEW Severity: normal Priority: P1 Component: iptables AssignedTo: laforge

Hi people! Do you guys know where to find a STUN Server Howto?! Why?! We all know, to get Asterisk behind an NAT Router to run, is a bit tricky, and you might have to fire a lot of holes in your firewall. However, I would appreciate it very much if somebody could give me great links of how to set up a STUN Server. Tamer

Dear all, I have a problem in writing a variable to a NetCDF-File. My code works pretty well until the step put.var.ncdf(): # Get variables #--------------------- data1 <- open.ncdf("PREC_me_03-1500.nc") prec1 <- get.var.ncdf(data1,"PRECT") dim.time <- get.var.ncdf(data1,"time2") close.ncdf(data1) # Calculation #--------------------- spi24.me <-

Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd

I have a laptop that hangs up on shutdown saying that at-spi-bus-launcher is still running. Since I have no use for at-spi-bus-launcher anyway, I would like to get rid of it, but attempting to remove the at-spi2-core rpm wants to remove 99% of my desktop as well. The only way that I can see to get rid of it is to make /usr/libexec/at-spi-bus-launcher non-executable, but that's a pretty

Le 01/05/2015 00:05, Andrew Martin a ?crit : > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> >> To: asterisk-users at lists.digium.com >> Sent: Thursday, April 30, 2015 4:43:33 PM >> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In >> >>> I am running Asterisk 11.12.0 on CentOS

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 05/05/2015 10:59 AM, Andrew Martin wrote: > > > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> To: >> asterisk-users at lists.digium.com Sent: Friday, May 1, 2015 6:42:38 >> AM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently >> Cannot Call In

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Thursday, April 30, 2015 4:43:33 PM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and > > internal phones are located on