similar to: FW: NAT router and off-premise SIP audio problem

Our network is connected to a cablemodem using a dynamic DNS service to resolve our address. The Asterisk server has been alternately set up behind a NAT router and without a NAT router -- that is, with two NICs, one of which is providing NAT to the rest of the network; the office SIPs are behind that with static private IP addresses. Off-premise SIPs are all behind simple NAT routers.

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

Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. We had slightly different parameters, e. g. that we would

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

Hi everybody, I finally want to get rid of 1-way audio problem. Please help me here. I have 3 scenarios. 1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is

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

Just in case other people on the list have this problem... Begin forwarded message: > From: "George Lin" <glin@cosini.com> > Date: Thu Aug 14, 2003 6:54:46 AM Europe/Budapest > To: "Paul Cheng" <asterisk@klarium.com> > Subject: RE: FW: [Asterisk-Users] SIP NAT question > > Dear Paul, > > Thanks for the suggestion. It works now. > >

Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

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

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

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

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

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 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

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

We had very similar problems to this which drove us insane for ages. Basically we use VoIP trunks (SIP) for all our inbound + outbound calls. Call quality was good however we would get random problems where people could not hear us or us hear them for about 5-10 seconds at a time. After weeks of trying to get to the bottom of the problem it appeared our VoIP trunk provider (sentiro/sip2go) had

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

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