similar to: Asterisk NAT

Hello, I have an Asterisk server with a public IP address and a bunch of clients. Most of my clients are behind NATs (sometimes two clients are behind the same NAT i.e in the same private network). I want to use ICE so that the clients behind the same NAT can send RTP traffic directly to each other and other clients use Asterisk or a TURN server. I tested a specific scenario using two Linphone 3.7

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

I have an asterisk server up and running, using Firefly in IAX mode works great, even with Firefly behind a NAT (as expected, since IAX works really well with NAT). Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from behind the NAT, and I can't seem to get there. At this point, the phone will successfully register with Asterisk, and the Asterisk qualify messages get

Hello I think I finally understood the issue/solution, but I'd like to make sure I'm correct: - In Diana Cionoiu's famous article on Freshmeat (http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol), regardless of whether SIP end-points use a public IP or are behind a NAT, RTP packets flow directly between the two SIP end-points because the SIP server only acts... as an

Hi, I am trying to run an Asterisk (1.4.10) server on Ubuntu Linux Fiesty Fawn. The server is behind NAT. I am testing SIP with the X-Lite client from xten. The client is also behind NAT. I realize that this is amongst the worst configurations, but I have been made to believe that it can work... eventually. However, currently SIP call set up seems to go fine, but no media is transferred in

This week, I've been really busy with the launch of a new Swedish Voip provider, www.bbtele.se, so I haven't been able to follow the Asterisk community and haven't been very responsive either. My apologies if you've tried to contact me and I did not reply quickly or at all. So to cover up (can't report on what is happening :-) I dedicate this issue of Asterisk Sunday News to

I currently do this for about 30 different cisco 79xx's connecting to some hosted Asterisk servers. Asterisk listens by default for any SIP connection on UDP port 5060. And will use RTP UDP port 10000 to 20000 The phones use UDP Port 5061 for incoming connections (from Asterisks or other SIP Devices) and use for RTP, UDP port 10000 to 20000. Now, if you are going to have the two remote

Hi, I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk for PSTN calling. Asterisk is configured to support nat with nat=yes in sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media using 183 Session Progress. So If I call a PSTN number which has IVR message played before the call is connected (via 183), those media RTP packets do not reach the

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

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.

Guys. Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of ways to get around nat but I would like to hear some success stories about handling nat users with multiple voip phones behind nat. I have my asterisk box behind but ports are forwarded (5060 5004 10000-20000 for rtp and 4569 for iax2) but still.. I can quite figure out what ser and stund have to do on this

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, A brief 6-step guide on how to hardcode a change in the Asterisk source that will allow it to work from behind a nat device. I know it?s messy, but it may prove useful to some people. 1. First punch a whole in your nat device. I just forwarded the port 5060 (for sip) and all ports between 10000 to 10020 (for rtp) to my asterisk gateway. 2. Now make sure your /etc/asterisk/rtp.conf correctly

Hi, I have been experimenting with NAT and Asterisk a bit now. Though I have made progress along the way, I have come across the following problem. I'll really appreciate if anyone can provide me any help or pointers. Thanks! Successful Scenario: ------------------- All sorts of NAT calls are successful with full two-way media when both end points are locally subscribed users. Problem

Hi Ray, I was wondering if the "qualify" option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being

Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

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 new to VOIP, Linux and Asterisk. Through a lot of reading (this list, voip-info.org, documentation, etc.), I successfully installed FC3 and * on a new Dell SC420 with two X100P connecting to two PSTN lines at my office. I've also installed AMP to help me configure IVRs, call groups, extensions, etc. I use a Handytone-286 ATA and x-lite clients on the internal network and all works

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either