search for: externalip

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK, but whenever I dial another S...

...y cisco 7970 remotely to it, but it just won't go. I am running 1.4.2 internally and the phone registers fine to it. I'm using the latest firmware (i think) - 8.2.1S On the server in question I have tried the following for the sip declaration: qualify=never nat=no (yes) defaultip=(natip)(externalip) md5secret=<md5pass> or secret=<secret> Nothing seems to work, and I continually get "sip 401 unauthorized" messages on the console when the phone tries to register. I've spent a number of hours on this googling and searching for anyone working with 1.2 and 7970's, b...

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

...g for several days, no errors/warnings. Simply added info for broadvoice and I can dial into my system via the remote, but dial out from my box renders the following: I get one ring on the remote line, then on the console chan_sip.c:675 retans_pkt: Maximum retries exceeded on call (calltracktxt)@externalip for seqno 102 (Critical Request) -- No one is available to answer at this time repeat of the above with a 102(Non-Critical Request) I am natting going out to the world with a small firewall appliance.

...setting the phone's internal IP to be the dmz. This allowed me to set up the phone using the web interface and now it registers. We had NAT problems so I set the NAT features of the phone: IP Address: 67.136.nnn.nnn Signalling Port : 5060 Media Port Start: 10000 In sip.conf, I have nat=yes externalip=67.136.nnn.nnn qualify=yes I can call the user and she can hear me. If she calls me, no voice can be heard either way. When I run sip show channels, I see: Peer User/ANR Call ID Seq (Tx/Rx) Format Last Msg 192.168.0.169 805 02fe2b2a684 00102/00000 g729 Tx:...

Hi All, I've an Asterisk Server behind a NAT. Using DNAT, I've opened port 5060 and all 10000:20000 udp. Sip configured with externalip and subnet. I've another site, also with NAT, where I map the rtp port (as defined in the client) to map to the local client (DNAT). Using Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always in the sip ext configuration but works quite well. However, late...

...N Dom and connected together with a IAX trunk. This setup allow us to use our both public IP (different ISP's) and to have failover solution in case of a problem on one of the ISP's line. Is it a way in 1.4 to do the same but with only one Asterisk instance? First problem we see is that externalip in sip.conf can be fixed for only one IP. Second is, if we could put 2 external IP's, if one packet is going out with, let's say PublicIP#1, and at this time the link from this ISP is down, packets will -helped by external routing- go out through the second line but never come back. Do...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, all! I'm being based on this document [1] to send and to receive calls using ekiga.net. But I'm seeing, in an Asterisk console, several messages of this type: [Feb 17 21:19:15] NOTICE[11875]: chan_sip.c:7715 sip_reg_timeout: -- Registration for 'danib2 at ekiga.net' timed out, trying again (Attempt #4775) -- Got SIP

I'm trying to move my Asterisk deployments under a Virtual IP address and now remember why I dislike this. My primary Asterisk system is now behind a firewall in private address space. My question is what ports are needed to be opened just for the purpose of placing outgoing calls. I would have assumed none, but I can't even get replies on registration from any of my 3 VoIP providers.

I'm a database noob, and it really seems like it would be overkill for my setup: I just want to proxy all connections from my DMZ to my internal mail server -- same internal server for all users. I used to use perdition for this set up, but am having issues getting it to play nicely with my new servers. Can I bypass all of the SQL stuff and just have dovecot forward all POP/IMAP

...imeout: Registration for '703XXXXXXX@147.135.8.129' timed out, trying again -- Got SIP response 404 "Not found" back from 147.135.8.129 Urgent handler My broadvoice config in sip.conf looks like: [general] context=incoming ; Default context for incoming calls externalip=82.41.201.XXX register => 703XXXXXXX:PASSWORD@147.135.8.129 port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; address to bind to (0.0.0.0 binds to all) srvlookup=no ; Enable DNS SRV lookups on outbound calls [B...

I've read a lot of the descriptions of handling NAT with Asterisk, and the use of both the nat and canreinvite flags. I am very familiar with Sip and NAT but have not seen an answer to the following question. My Asterisk server runs on a machine with two ethernets. One is an external net, with exposed IP addresses. The other is an internal net with natted IP addresses. Thus the server

...isk on the INSIDE of a NAT device If you have Asterisk on the INSIDE of a NAT device, trying to communicate with a SIP proxy (like Free World Dialup) on the outside or having phones on the outside, you're in another ball game. Check the sip.conf.sample in the Asterisk configs directory for the externalip and localnet settings, that tells Asterisk what the outside IP address is. And yes, we could use a STUN client in Asterisk so that Asterisk could figure out what to do. Any coders? You're wanted. If you have phones on the outside, port forwarding is your best friend. As you can imagine, this i...

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be

...I then built a few cheap x86 boxes to simply collate share information then remote browse sync to the DMB in the main office. It worked! Network neighborhood was never happier. Since then my knowledge of iptables has increased and I now invoke `/sbin/iptables -t nat -A POSTROUTING -o ipsec0 -s $EXTERNALIP -j SNAT --to $INTERNALIP` This allows me to connect to services on remote internal networks from an IPsec Gateway. AHA I thought. I can eliminate those unsightly pentium samba boxes and place the share collation on the reliable (and underworked) firewall/IPsec Gateways now that I could actuall...

Hi, first i must tell you, that my english is not the best, i hav learned my english from manpages and documentation. Please excuse this. I have setted up a Box w/FreeBSD 4.7-RELEASE for connecting to the w3 through an DSL/ATM-Connection. Now i know the stateful handling of firewall-rules under linux with iptables.In the second i have understand that FreeBSD comes with the netfilter-extensions.

(If this is a double post, I apologize, my email client crashed when I first sent it) I need some help to configure a secure network on my Xen server. I have been looking online and it seems a I need a routed network. But I am having a terrible time implementing it. My setup: Xen 3.4.2 CentOS 5.5 Dom0 1 NIC (eth0) All guests will be HVM What I want to do is something similar to a firewall