similar to: Grandstreams+Nat

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

I have a few questions regarding SIP and NAT that you may be able to answer. In both cases, I'm "assuming" that the customer will use SNOM phones and/or xten soft-phones. Q1: I know that it is possible to use a STUN server to handle SIP over NAT. Does this require any special configuration of the NAT router? For example, will I need to configure port forwarding? Q2: If I know

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

Guys. Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of ways to get around nat but I would like to hear some success stories about handling nat users with multiple voip phones behind nat. I have my asterisk box behind but ports are forwarded (5060 5004 10000-20000 for rtp and 4569 for iax2) but still.. I can quite figure out what ser and stund have to do on this

Hello, I have a problem with a Grandstream being behind a symmetric nat. The box which does the nat is a german "Fritz Box". This one does nat for the internal network. In the internal network is a Granstream BudgeTone 100. The nat router has a dial-up connection, so ip changes on every dial-in. |------------| |------------| |--------| |Grandstream

Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

Good day all I have asterisk on a box with one network card I have a 2 companies setup on the system. To keep all apart I binded a different ip to the interface,i,o,w eth0 192.168.0.254 and eth0:1 192.168.1.254 And in sip.conf I took the bind setting out So each company's phones are on a different ip range,and all worked well So we decide to pull the snom190 out and exchange it with a snom360

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When

Has anyone made this work? For me everything is fine until I switch canreinvite form no to yes. What happens is that asterisk hangs on "attempting native bridge" ... from what I understand "attempting native bridge" means that the RTP is routed through asterisk (just without any codec translation) But it shouldn't do that ... right? ... canreinvite is set to yes ...

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

Hi, I install * and is working fine. I have 3 FXO cards w/ 3 phone lines. All the phones are SIP phones (Grandstream). The SIP phones from the same LAN w/ Asterisk are working but on the external phones (from the Internet) I don?t have sound. All the Grandstream phones from the Internet are register from different locations behind a NAT. All the sip users are register on * but the main issue is

I have used Grandstream phones for years, and have just started testing a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling and don't know whether it's just a limitation or something I haven't done correctly. The Asterisk server is directly on the Internet with a public IP. The phones are on a private LAN with a NAT router to the Internet. The sip.conf entries for

I try to get H323 to run, but have so far only partial success: There is a Gatekeeper GK, where asterisk connects to. The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper. From the Network on the GK, asterisk is reachable via the number 070333333. I have an extension on asterisk 6002, which is reachable. I try to call a number attached to the gatekeeper (070168177) with the

Good day all I added the music on hold entry in vpb.conf and commented out default line in musiconhold.conf. Asterisk starts up with the default mp3 but as soon as I remove it and add my mp3 it just doenst start up and gives a broken pipe error? Please Help or advice Thanks ALtus

Good day all I'm just wondering for interest sake I have a box,hostname=myname.co.za,running sendmail If I send mail to someuser@myname.co.za it try to deliver to the box,witch does not have the mail box.How do I tell sendmail that it should send mail to myname.co.za's mailserver. I know how easy it is to change the name but there's a lot of reasons why we can.It is not in the

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

Hi, I noticed some issues with how grandstream handles stun test. GS is running version 1.0.4.50. First I reset the NAT router. Then reboot GS, get results of "restricted cone". Immediately reboot GS, get results "full cone". I tried quite a few public and commercial stun servers. Also tried different model/version of linksys routers. I always got the same issue. Winstun on

I'm at my wit's end! I've spent 2 days now trying to get what I thought was a very simply SIP + NAT arrangement working. I've trawled the web and picked brains, but nothing anyone suggests work. My setup is very simple. I have a * server in a datacentre, with a public IP address. There is no firewall in place, it's completely open (at least, as far as I'm concerned). I

PH> I am setting up a snom 360, and the lights come on OK when the mapped PH> user makes an outgoing call, but when the user takes an incoming call PH> the light does not come on. PH> I do not want to install the bristuff patch if possible. PH> (although I can see that with the devstate command I can make the lights PH> do whatever I want) First, ensure that the 360 has