similar to: Traversing the NAT

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

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

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

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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

Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The hostname

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

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

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

----- 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 > > Le 01/05/2015 00:05, Andrew Martin a ?crit : > > ----- Original Message ----- > >> From:

Hi, I've been experimenting with an IPv4 and IPv6 VoIP setup using SER. I'm using Asterisk for voicemail, etc. but as this only works for IPv4, I had to do a number of tricks to get it going for IPv6 phones. I was wondering whether there is any interest or plans in the pipeline to have Asterisk IPv6-enabled. Any info, especially by the developers out there, would be welcome. Thanks, --

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

One of my clients today asked me about TLS support for encryption of SIP payloads, and I didn't have an adequate answer as to why it wasn't supported or even discussed. Some archive searching finds scant mention of this in reference to Asterisk. Of course, encrypting the SIP payload is only 1/2 the problem; the payload itself is the next problem. I understand that IAX solves these

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

Hi, I''m playing with the builders in libxc. I am having trouble debugging the thing. I was wondering what options I have. - When I run ''xm create'', what process calls down to the xc_builder stuff? Can I attach to it with gdb? - Is there any way to spit out debug output (printf?)? Where would it go? - If I make changes and do a make install in

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",