similar to: OK to have Asterisk and clients behind firewalls?

Hi, I have the following problem. I have 5 phones behind the same nat (canreinvite=yes). it works fine to receive calls and to make calls. sound quality is good, so everything works fine. The poblem is that the phone behind nat cant call each other. It works if canreinvite=no. But i want to do this. Does anyone have an idea? Regards, cjk.

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

Hi, I'm having a problem with my asterisk, trying to connect to a CISCO 2840 IOS12.x ASterisk is behind firewall NATing, when it do the handshaking for RTP, it sends his internal IP instead of sending the external one. How can I tell the asterisk box, to modify that and send the external IP? I tryied with Sip.conf's externip=xxxx and localnet=xxxx, nat=yes Nothing seems to change the

Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the

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

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

Hi, I have following setup: BT100 ---- Firewall/nat 1 (www.ipcop.org) ---- Internet ----Firewall/nat2 (Vigor) ---- Asterisk . I'd like to use BT100 as local extension to Asterisk. I've done simple setup and BT100 can call Asterisk and place outgoing calls. However I cannot set him to qualify, cause it is claimed as unreachable. I have port redirection at Firewall 1 (to 5060 and rtp

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

Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

i have this configuration , An Asterisk server connected to my private LAN 192.168.10.0/24 when i do port forwarding for port 5060 so that i make a call from Internet into Asterisk wireshark show the message "destintion port unrechable" i configured sip.conf for "nat=yes" and "qualify=yes" and "externip="my public IP" did i forget some other ports

Hi all, I have set up Asterisk and I am able to register with my SIP provider and receive calls. When I try to register with Asterisk from outside I can place calls but tthe other person can't hear me. Have opened port 5060 UDP as well as port 10000 to 20000 UDP. Any ideas? Thanks, Christian

Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the "fixups" disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the

Excuse my ignorance but if i have an asterisk in a LAN, and i have users in their homes/internet (dozens), in order to correctly connect those users across my firewall, what is the technology that i need to buy, called? secure border gateway? session controller? secure gateway? the audiocodes site seems to have many names for the same thing...but i better ask here and learn before i make a big

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.

I'm sure you all already know this, but it's generally better to have the firewall be a separate physical machine from any server or client machines. Among other reasons, if an attacker can exploit an application running on the firewall machine, the efficacy of the firewall is compromised. You should dedicate a hardened, minimally configured machine for firewall use. If you have lots of

Hello, is it possible to run tinc behind an unknown firewall(simple NAT) e.g. on windows clients to join an tinc network? These firewalls are not maintained by me and not possible to modify. Thanks for help Kai Tinc: Discussion list about the tinc VPN daemon Archive: http://mail.nl.linux.org/lists/ Tinc site: http://tinc.nl.linux.org/

I've searched the archives a bit and have not really come up with a good answer to my queries. I have * running on a RH9 box behind a LinkSys NAT box. I can talk with iConnectHere outbound just fine. I am trying to configure an inbound Xten softphone from outside. I have that user set as NAT in sip.conf (seems to help), but I still cannot establish a full session. I think the problem comes

Hi All; I succeeded to have a success call from Polycom behind NAT while Asterisk has public IP address, but I was not able to have a succeed call (it was established, but no voice running, and then the call disconnected) if Asterisk behind NAT and Polycom behind NAT. When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to asterisk (at asterisk router) and to Polycom IP