similar to: Asterisk both behind a NAT and outside at the same time

I decided to give the whole family IP phones for christmas, all hooked into my asterisk server, so all the nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly

Hello everyone and welcome to my first post to the list! After studying for a couple of weeks, I finally built * for the first time last night, and of course had the same SIP-behind-NAT woes that plague all of us who use NATted connections. It was therefore with no small joy that I read the fix for that that Walter Snel proposed (q.v.:

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

We all love Asterisk's ability to Dial(chan1&chan2) and take the first that answers. However, I have been encountering a problem when one of the channels is an external phone behind NAT and another is a local phone on the same net as the asterisk server. All have canreinvite=yes, and the phone behind NAT is correctly using Stun to give its external ports, which are opened to it in the

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

--- Peter Zeltins <peter@fintrading.com> wrote: > > > Well, I happen to be one of those very specific cases... ;) and looks > like > will have experiment with it myself. Although I'd hate to re-invent > the > wheel. > > Peter Checking e-mail this morning it looks like we have two independent "fixes" that both do what has been suggested in this

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, 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.

I know that NAT is something no one really likes to talk about, but does anyone know how work with it elegantly? There are many providers which deal with it on a daily basis in fact they cater to it, is this possible to do with asterisk or does it require other exotic setups? I even know of a provider which uses asterisk with many different types of devices, and they handle all NAT config on

Hi folks, Ok, I've seen this question go unanswered on the mailing list, and I assume it's because no one had the heart to break the bad news to the guy asking, but be honest with me, I can take it. At this time it's flat impossible to have multiple IAX phones behind a NAT without using an * gateway because there's no way to have a client listen on a port besides 4569. Is

Hi Guus, Am Freitag, den 25.09.2015, 09:36 +0200 schrieb Guus Sliepen: > On Fri, Sep 25, 2015 at 08:41:06AM +0200, Marcus Schopen wrote: > > > I'm running some tinc clients behind a NAT (masquerading, Cisco Router) > > connecting to a host outside on a public IP in a different network. The > > tunnels get unstable every few minutes and I see packet loss when > >

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

I am trying to get a SPA-3000 to work behind NAT - for the sake of the exercice. The SPA is on the local network at the address 192.168.0.125 behind a NATted linux router. The machine I am trying to work with is a friend's (let's call it lolo.dyndns.org) and I've installed Asterisk 1.0.3 on it. I can see the SPA register but when I try to make an outbound call I get the message:

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

Hi All, I'm having a really weird can reinvite issue. I've been banging my head around on this for days now.. I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone at 172.20.2.80 via server

Hi Guus, Am Freitag, den 25.09.2015, 17:04 +0200 schrieb Guus Sliepen: > Ok, that means by default the UDP NAT timeout on the Cisco is extremely > short. > > > I check the manual of the the Cisco NAT for any TCP/UDP > > timeout settings, but there is no way to modify anything like "keeps > > TCP/UDP connections alive". > > It wouldn't be called

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 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

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph