similar to: *** Asterisk Sunday News: The SIP NAT Special

Spring is back in the Stockholm area. After a few day's worth winter re-runs, the sun is back and night-time temperature is at least 5 degrees celsius. Time to move out all my annual flowers and prepare the garden for summer. Sweden is famous for our annual five week holidays - by law. From june to late august, it's almost impossible to make any business decision, since there's always

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

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on

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 In case Asterisk is used in a private LAN behind a firewall while allowing remote SIP clients to connect from the Net, we must open UDP5060 for SIP and a range of UDP ports (as set in rtp.conf) so let incoming voice packets. Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP...

I think I have got * worked out as far as getting users on a small private network talking with each other, but when it comes to the bigger picture about talking between private networks connected by the Internet then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc. Before I start let me make it clear that I am not looking to drop out onto the public telco network anywhere, not at

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

On Wed, Nov 11, 2015 at 09:04:20PM +0000, Etienne Dechamps wrote: > Unfortunately, I often notice that even with the advanced UDP hole > punching mechanisms that tinc uses, I still come across cases on my > network where two nodes behind NATs cannot talk to each other because > one of the nodes is behind a "strict" NAT [...]. This type of NAT > will stop any UDP hole

fauno <fauno at kiwwwi.com.ar> writes: > Etienne Dechamps <etienne at edechamps.fr> writes: >> (I realize that this means UPnP support could possibly be achieved >> simply by suggesting that the user spawn some standalone UPnP client >> process in the background from the tinc-up hook. That's not very >> user-friendly, though. Especially on Windows.) >

One for the gurus.. I have seen there has been a lot of discussion about using SER with Asterisk.. This to me seemed like an over kill becasue it would basically be doing most of what Asterisk is doing anyway unless you create some weird and wonderful config in SER.. Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

Good day all I cant get my grandstream bt-100 to register My asterisk is on a public ip and the phone behind a nat firewall I added nat=yes in sip.conf and did this on my grandstream set the GS to "SIP server=asterisk.yourhost.com" and leave "Outbound Proxy" empty * set the GS to SIP port 5060 and RTP port 5004 (and "Use random port=No") * set the GS to

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

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 have a problem with a Grandstream being behind a symmetric nat. The box which does the nat is a german "Fritz Box". This one does nat for the internal network. In the internal network is a Granstream BudgeTone 100. The nat router has a dial-up connection, so ip changes on every dial-in. |------------| |------------| |--------| |Grandstream

> What machine are you running (namely what OS)? Debian. > I dont understand your approach. > Why running a 'streamer' behind a nat? > Not enough 'resources' to rent/ rent to buy a ded. Server? > Mean, can't expect to satisfy a lot of listeners this way. :-) I am listening the Muazkhan indications XD: > > Hi Muaz Khan, >> We are adtlantida.tv and

Hi, I recently encountered a weired situation where my budgetone stopped working. My network is like this: Asterisk on Public IP ----------- ADSL NAT Router ----- GS01, GS02, GS03 on Internal IP We have an Asterisk server running with a public IP address, which serves as the master PBX. On a remote site, we have 3 Budgetones all having internal IP addresses assigned by the ADSL NAT router. The