similar to: Asterisk behind Multi-NAT question

I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE

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

Hi, We're installing an * box next week (pbxtra from fonality) and I'm trying to come up with a solution for remote users that want a phone in their home. I need VPN and QoS capability, wireless support would be a nice to have. Ethernet handoff is fine, i don't need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces

Hi All, How to park and pickup a call? The scenario of park and pickup described as below. UserA made a call to UserB, and the call ware connected, Then UserB parked (or hold) the call, and told UserC to pickup the call on one line, and then, UserC pressed some keys to pickup the call. Who can tell me what's the Park/Pickup's typical flow in the Asterisk. And how to set the sip.conf,

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

Hi all, I finally got my IPSec tunnel from my Fedora firewall system (running Shorewall 4.0.6) to a remote Draytek Router up-and-running, but I''m having difficulties directing traffic through the tunnel. From the output of "racoon -F -f racoon.conf" and the connection status page of the Draytek I can tell the tunnel is UP, but ping and traceroute requests to several hosts

> From: George Lin [mailto:glin@cosini.com] > > I want to deploy multiple SIPs phone in our office. And we have shutdown > the > firewall at our office router(with ip 211.x.x.x). we have deployed the > asterisk with IP 218.x.x.x. > > All SIP phones have 192.x.x.x. We have something similar George, * sits outside the firewall with a registered IP address, the SIP phones

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

One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they

Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried

Hi, I have just got my SNOM 200 to start doing some real testing with *.. I am trying to use the GSM codec but the quality is really bad, Is that normal? does anyone actually use GSM?? Also are there any 'gotcha's' that I need to look out for so I don't spend hours trying to get somthing working that really doesn't work anyway.. Thanks.. later.. --

So I thought I had SIP and NAT cracked a long time ago, but something's just happened that's sort of upset the cart )-: I have an * box behind a NAT firewall. Nothing unusual there, this is something I've done many times - sip.conf has the correct nat= localnet= externip= settings, the router has ports 5060-5069 and 10000-20000 forwarded to the internal IP address of the

Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up" repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: < Unnumbered frame: < SAPI:

hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards

I'm in conversation with Draytek's pre-sales dept.............. Here's the most recent reply: <Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards,> All I want to know is, if I buy one of these routers, will it break my setup or

Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and

Hi, I have setup a IPSEC VPN using Openswan to connect a Draytek router to a CentOS 5.2/Shorewall 4.2.9 firewall. The VPN establishes OK but I''m getting a problem with packets from the left hand subnet getting masqueraded rather than routed down the IPSEC VPN as though they were going out onto the net. I''ve spent the last day searching Google and so far I''ve hit a

Hai Peter, Chipping in here. > > > Hi Rowland, > > You are right about firewall boxes. At least Cisco ASA is a terribly > (over) complicated device. People who are not Cisco pros should be > warned. Stay away, you will just waste your time, get frustrated, and > get sleepless nights. > > I don't blame the Cisco ASA here. In my case, I hadn't much

I have a DrayTek Vigor3300V gateway with 8 FXO ports. I am trying to configure asterisk to dial out on the gateway. I have one of the FXO ports configured on sip account 100. If I dial the sip account then the router gives me dial tone, with which I can dial a number. Unfortunately this is not the behaviour I desire. I want to setup the FXO port as a trunk with out it giving me a dial tone. I have

On 10 February 2015 at 12:55, jg <webaccounts173 at jgoettgens.de> wrote: > >> Some firewalls have a 'consistent NAT' option that needs to be enabled, >> otherwise you get the symptoms described. >> >> While reading about NAT, I came across this web site: > http://nattest.net.in.tum.de/ > The test tool looks at various NAT related properties and