similar to: Asterisk to sip client behind Firewall/NAT - can call but cannot receive calls ?

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

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

Hi, I recently encountered a weired situation where my budgetone stopped working. My network is like this: Asterisk on Public IP ----------- ADSL NAT Router ----- GS01, GS02, GS03 on Internal IP We have an Asterisk server running with a public IP address, which serves as the master PBX. On a remote site, we have 3 Budgetones all having internal IP addresses assigned by the ADSL NAT router. The

I think it should work at least for TUN virtual interface as TUn works at IP level. This is a sample configuration. firewall1 lan = 172.16.1.11/19 (ALWAYS ACTIVE) - "Physical Network Interface" - system config as ifcfg-... 172.16.1.10/19 (VIP Keepalived Make active) - Active/Passive configuration with firewall2 firewall1 vpndr1

Hi, I wonder what does this warning 399 mean and how to workaround? "sip show peers" says that sip client is unreachable althought it works with some eexceptions ... I saw posts in this list about setting codec to ilbc, is this right action ? Also, I'm very interested if anyone succeded on Asterisk behind one firewall and Grandstream behing another ? I' almost got it working

This is a vpn for Disater Recovery sites, so it is not necessary to have a seamless failover, strictly speaking. Encryption instead is mandatory. Testing we found that on Keepalived failover remote Tinc take few seconds to reset the connection and correctly re-connect to the new active firewall (probably new firewall resetting the connection + PingTimeout + some seconds to reconnect). This is

This is what I want to avoid :D I want an active Tinc virtual interface active with ip identical of the other firewall, without ip conflict on the same network. Do you know if Tun type virtual interface on one host can have same ip address of another host in the same network without ip conflict ? ie if a tun virtual interface can work active without transmitting on real network ? or if such a

Hi Saverio, I found conflict: 172.16.1.10 00:50:56:1b:ba:5e VMware, Inc. 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 2) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 3) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 4) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 5) So my assumptions were wrong ! :D Probably Virtual

Hi To investigate the UNREACHABLE issue I'm having, I need to have confirmation that it's OK for the Asterisk server to be behind a NAT router, and also have clients elsewhere on the Net behind their own NAT router? I know that clients must use STUN to resolve their public IP and punch UDP holes in their firewall, but is there something special that must be done in the configuration

Hello may situation is the next: Asterisk <--> NAT1 (router)<---> internet <--> NAT2 (router) <--> x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone

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

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

I just tried to set up a tinc test installation, but I guess I did something wrong, because I can only ping one direction. Here's my setup (lan1,192.168.99.0)<-->firewall1<-vpn->firewall2<-->(lan2,192.168.100.0) firewall1 has tap0 on 192.168.88.2 firewall2 has tap0 on 192.168.88.3 i set up the following routes : on firewall1 : 192.168.100.0 gw firewall2(192.168.88.3) on

Hi list, I try to install the following network: /-----------------\ | Firewall 1 | \-----------------/ ___________________| |_________________ | 10.2.6.1 10.2.2.1 | | eth1 | | 10.2.6.11

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,

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,

Samba version 3.0.9 on Red Hat 9.0 workstations: NT 4.0 SP6 When I took over this project, there were two domains connected by a VPN. For reasons that I'm unable to understand, some new users wound up with the intended domain of SATA and some wound up in SATB (even though they joined the SATA domain. Recently, all kinds of problems have materialized and the only thing I can find wrong is

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

Hi. I need connect some LAN stations with SJphone to an Asterisk Server published on Internet. My Lan Clients access to Internet using a small linux firewall/proxy server. I use the next firewall script. That is a simple script with default policy ACCEPT, and NAT to share Internet. I can connect to the asterisk server, authtenticate the users in the server, and dial to any extension, but

I have a couple of Budgetones that I am playing with trying to get them to work with * from a remote network over the Internet (yes NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded (UDP) to my public address (through a Cisco PIX). Internally, I can setup my budgetone, it registers and works great. I then have a Linksys router connected to another Internet