similar to: Asterisk + SIP + NAT - seriously, what's the secret?

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've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems. However, I'm about to deploy about 5 phones (either budgetone or GXP2000's) all on a LAN behind a NAT- on a different network than the Asterisk server. Should I look into using STUN servers? Will this setup be a

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometimes it doesn't, etc. I've come to the same conclusion as the wiki: it's probably better to avoid this

hi, I have the following configuration: Grandstream --> NAT (Netgear RP614)-->Internet-->Asterisk(public IP) i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable my sip.conf file is as follows:

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 a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS about a week ago. My problem is that I'm only getting half-duplex communication -- I can hear voice from the Asterisk server but the server does not understand any voice from

On my Grandstream GXP 2010 I have the possibility for 6 channels and thus 6 different accounts... Line 1 I define an account that registers directly to an online Asterisk-server, somewhere in a datacentre. Line 2 I define an account that registers to the local Asterisk-server (NSLU2 unslung) When I activate both accounts, only the first account (to the Asterisk-server on the internet) registers.

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

Greetings list, I'm coming across an issue with some of the GXP-2000 phones we have out in the wild at clients' employees' homes. In most cases they're behind consumer ADSL NAT routers on a dynamic IP from their ISP. In a nutshell, the phone is unable to be called unless it's restarted first, after which it's fine for a good few hours, then it stops working until

I have 2 Grandstream HT-286 devices and an Asterisk server. The * Server is not using NAT and has port 5060 opened up. One HT-286 is using traditional NAT and the other HT-286 is behind a residential DSL router/firewall. I have the HT-286 setup as the "DMZ Host" in the router/firewall so that all incoming connections are forwarded to the HT-286. HT-286-1 ====== NAT FW ====== * Server

We are interested in how other people are handling NAT problems. We have several customers all of which have some sort of firewall/NAT device at their location. For simplicity sake, all customers' internal networks are 192.168.*.*. Our asterisk box is on public IP not blocked by any FW/NAT. I use QUALIFY=yes on all our customers' phones and I feel that sending out 80-something

Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in "general" that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network

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",

At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: >>[snip] >Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk >can handled the NAT traversal all by itself with Qualify (as John points >out) disabling the NOTIFY will not change anything. > >The NOTIFY will in no way affect the status - unreachable/reachable. > >Another problem with the SIPURA is

Hello, We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. It is probably the problem with port forwarding on router - but I am not sure how can I forward same sip ports (5004 to 5100) to two phones (nat addresses?)? Any help appreciated! Z. Zivanovic -------------- next part

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

I'm trying to understand IAX and whether or not it would solve my difficulties: 'The primary goals for IAX were to minimize bandwidth used in media transmissions, with particular attention drawn to control and individual voice calls, and to provide native support for NAT (Network Address Translation) transparency. Another goal is to be easy to use behind firewalls.'

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