similar to: Multiple SIP phones behind a Linksys firewall

<asterisk-users at lists.digium.com> Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall Date : Sat, 2 Feb 2008 18:25:16 -0700 > And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall? > > In terms of nat and Cisco 7960s I've never had a

Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI "binding" improved since then? 2) Is there any chance of a "real" API for Perl? Thanks much! -Ken -- This message has been scanned for viruses and dangerous

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

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

Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they

Hello List, Can you please point me to the right resources on making multiple sip phones behind a firewall w/ private address work with asterisk w/c is on a public network. I have seen STUN on the grandstream and Xtunnels on X-lite. What is most deployed by members here with similar setups? Thanks. -- Cheers, Paul P. Pongco

Hi, When we use BudgeTone-100 in our Intranet together with our Asterisk IP PBX everything is working OK. When we try to use the phone behind the Firewall we can't do the connection. When I try to use STUN Server: 128.107.250.38 there is no result. The only way in which I have audio from the one direction (BT-100 to Asterisk) is when I leave blank STUN Server and specify the IP Address in

Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside

Hi Everyone. I'm in a quandry & don't know which way to go. - Obviously I'm an Asterisk newbie although I've been watching this list for over 2 years now. I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running here at home. - It's on my home LAN - NAT'ed behind my LinkSys router. - On the same LAN I've got a Cisco 7940, 7960, and

<MOD NOTE:Please kill/bounce my other email, it was accidental.> I just pulled down the newest CVS and recompiled. FWD (free world dialup) works now from *, and I AM behind a NAT. I've nearly given up on the xten lite, iaxcomm sounds better. I'll be trying the other win app thats up-and-coming on the list later. It seems to have broken iptel, but that's not as important to

Has anyone successfully configured an Asterisk box which is behind a (NAT) router for Sipgate? I have mine behind a Linksys router and can successfully register and apparently call out but I get no incoming audio and can't be called from the outside. Any help would be greatly appreciated. Roger -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an asterisk server at home. I'd like to test one just installed elsewhere. Both servers are behind firewalls. I can see the session start in CLI, my congratulations is apparently playing and RTP is being sent. Hearing no audio. Can send key presses and see audio playing changed. "Peer audio RTP is at port 198.145.28.177:10180", but that never shows at the client side, behind

hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no

Greetings, I am trying to get an IP phone working through a linux based iptables firewall. I have an asterisk server with a public IP address. I ran netcheck from FWD. It says that it is a Port Restricted Nat. I tried the recommended FWD approach, changing the FWD-specific settings to the * server's. I have tried every conveivable config on the phone (Yes to NAT traversal with STUN

Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any "STUN Server:" setting in "SIP" tab. On Asterisk I think I only need to make changes in sip.conf isn't it? What

Hi, On my linksys/sipura phones/ATA, there is a setting called "NAT Mapping Enable" and another called "NAT Keep Alive Enable" These settings must be on in my setup so that my phones/ATA remain connected to my * server. My setup is: Home LAN - Pfsense (NAT, Dynamic Public IP)- Internet - PFsense (1-to-1 NAT, Static public IP) - Asterisk server. I was wondering:

Note that the newish SPA962 has 6 appearances and a color screen. I've noticed that the bright color screen does impress people when they first see it. PoE is also very nice and web provisioning was quite easy. I've yet to try a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

I have thew same problem. Ui tried with dyn dns in the externip field in sip.conf but I think the Asterisk does not allow this. Unfortunally I have every day a new ip. Maybe I can write a script witch takes my actual ip from externat and put it into the externip field. Maybe this solves the problem. -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com