similar to: looking for input on broadband router with QoS andVPN support

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

I have a SIP trunk to Broadvoice. My Asterisk box (1.4.13) is on public addresses behind a firewall. Originally it was behind a Linksys WRT54G running sveasoft. Sveasoft really can't NOT do NAT even when you turn it off. My Asterisk box is defined as the DMZ box to Sveasoft and it seemed like it was leaving all packets alone. Now I switch to a Centos-based firewall configured with

Hello, if you have a Draytek SIP phone, please check if the phone doesn't flood your server with SIP REGISTER messages. Draytek phones are broken and keep sending REGISTER messages after receiving 200 OK (even if expires value is long enough). Several such phones are flooding iptel.org public servers these days. If you have direct contact to Draytek developers, please send it to me.

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

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

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

As seen on my post at: http://www.sveasoft.com/modules/phpBB2/viewtopic.php?p=28112#28112 This works very well... It does NOT work with stable 4.0! sveasoft will be issuing a bug fix for this (4.1) in the near future. Final Rev of working script w/ asterisk support I'm not going to run alchemy on production machines until it is stablish. Remember to set your uplink properly and to set

Hi, This is a thank you message for all that helped me including Max from www.asterisk-support.ru with whishes of a Happy New Year. Althought I still have a problem I'm happier I've 50% of my task complete. I'm using two TA from Draytek (router 2600V / router 2500V) 3 ADSL lines (2 for TA 1 for ASTERISK with a Draytek 2500 no voice model). If I call from one to another (using

current setup SIP phone 192.168.1.30 --> linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)--> (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address.... or #2 asterisk trying to get back to me as 192.168 on public internet.. got

Greetings list, Thanks to all who replied to my thread a few days ago "SIP devices with packet loss tolerance". One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which

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

Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? Thanks..

Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing

Greetings all- I've got a rather odd situation and would like to know if anyone can shed some light on the issue. Some background- I've got an * system running 1.4.11 (yes I know it's older.. upgrades are planned at some point...). I also have a remote user with a cordless phone connected to a Draytek ATA device. When this user is on a call and receives another call via call

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the Digium web site said something about analogue cards being able to support "provider T1

Had something recently on 2 separate sites where there was a lot of audio lag on a call - and by a lot I'm talking 5 seconds or so. Two different sites, but the setup is similar: SIPphone A <internet> Asterisk <IAX/Internet> Asterisk <-> SIPphoneB If the phone called extensions local to it's asterisk there was no lag, but if it called out (separate IAX trunk) or to

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

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