similar to: Draytek Vigor 2900 & Asterisk

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

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 have recently installed a test Asterisk server but am having problems getting this to work with a Vigor 2600V. I have no issues using X-Lite to connect to Asterisk but if I configure the 2600V, it registers correctly but I don't get any sound at all although calls do seem to connect. The 2600V works fine with FWD and other services so I suspect I am missing a setting but as a user have

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 have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works

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.

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

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

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

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, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

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..

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

Hi, I'm looking for opinions on the "best value" router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source

My apologies for asking what is doubtless a dumb question, but I have scant experience in R. It would be very convenient in doing lots of plots to be able to do them in a loop that stepped through a vector of variable names. For example one could say x<-("mydates") y<-c("foo1","foo2","foo3") #where "foon" were vectors

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Greetings, I need to develop a way to manage the installation and update of R packages under linux (CentOS), hopefully using Puppet. I''ve got most of the rest of hodge-podge of install scripts and maintenance scripts translated into Puppet and need to tackle this beast now. R is a programming language for statistics, basically, and has a few jillion add on package libraries, very much

Hi, yesterday evening I tried to upgrade my Ultra 60 to 2009.06 from SXCE snv_98. I can''t use AI Installer because OpenPROM is version 3.27. So I built IPS from source, then created a zpool on a spare drive and installed OS 2006.06 on it To make the disk bootable I used: installboot -F zfs /usr/platform/`uname -i`/lib/fs/zfs/bootblk /dev/rdsk/c0t1d0s0 using the executable from my new