similar to: Thanks for help - Almost done - 50% - Can hear

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

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.

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

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

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! Are you also getting in trouble while trying to register in Broadvoice? Cumprimentos / Best regards, Helder Rog?rio __________________________________________ Microrede - Tecnologias de Informa??o, Ltd. http://www.microrede.pt *** ? There are only two types of people in the world, those who have lost data and those who will. ? -- Richard Nixon

Bob, Have you looked at any of the products by Zyxel? With QOS, VPN & wireless support they have: For ADSL: Prestige 652HW Firewall/Router: Zywall 10W & 30W I'll be honest, I havn't used any of these yet. We were looking for similar products to suuport our VOIP installs. We just ordered some demo units from Zyxel, we shoul have them later this week. I'll let you know

Hi! Is there a way to avoid being "at the middle" of communications between two SIP endpoints? So that we can avoid loosing bandwidth with it? Is there a way to "forward" the authentication to a IAX provider and "transfer" the call to it, avoiding using my own bandwidth? I've tested it with SER with some results, I was wondering if it is possible with

Hi, I've tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s

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..

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

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

Hello everybody, I'd like to have LPK (or something like that - getting public keys from LDAP) integrated into mainline OpenSSH. *** First of all, a summary. The project page at http://code.google.com/p/openssh-lpk/ mentions that a few distributions include LPK per default; but reading the various threads at Support for merging LPK and hpn-ssh into mainline openssh?

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 Centos linux 5.11 server with samba3.6.23-6.el5 with workgroup (not domain) This system has been running for four years with no real issues. There are 9 xp clients all within the workgroup, rather than connected to the domain. The server is used for windows shares and for openwebmail This week I had a problem with the draytek 2820 adsl router. I had to reset it and update to the latest 2014