search for: vovida

Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The h...

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a "vi" guy with CLI preferences. Gastman han...

...itself is the next problem. I understand that IAX solves these issues, but it will be some time before IAX is a "standard" on equipment. Does anyone here use hardphones or softphones that support TLS? Is TLS stillborn? Speaking of payload encryption, has anyone ever peeked at the Vovida SRTP libraries? They're free (as in, BSD-licensed) and even in C (not C++) and found on http://www.vovida.org/protocols/downloads/srtp/index.html Asterisk is sometimes the chicken, sometimes the egg - maybe putting TLS and SRTP into Asterisk will cause some more vendors to start putting...

...a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX <http://www.voip-info.org/wiki/view/sipX> from Sipfoundry <http://www.voip-info.org/wiki/view/SIPfoundry> is a native SIP proxy but also a complete SIP PBX * OpenSER <http://www.voip-info.org/wiki/vi...

...co phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to appear..... ;-) JT >Reply-To: <davidk@cisco.com> >From: "David Kelly" <davidk@cisco.com> >To: "Chok Lam" <chok@cisco.com>, "Vocal@Vovida. Org" <vocal@vovida.org> >Subject: RE: [Vocal] Question about Cisco IP hard phones >Date: Mon, 14 Jul 2003 11:56:45 -0700 > >Folks, > >For the time being, the low-end Cisco IP phones, 7902G and 7912G >support SCCP only. The 7905G supports both H.323 and SCCP, but &...

any way the you can get * to send a NOTIFY SIP message to all SIP phones? to have the SIP sets recheck thier configs etc?? Like this? NOTIFY sip:sip@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=1 Via: SIP/2.0/UDP ipaddress From: <sip:webadim@192.168.0.1> To: <sip:sip@192.168.0.3> Event: check-sync Date: Mon, 10 Jul 2000 16:28:53 -0700 Call-ID: test@192.168.0.1

Hi, I run a local exchange server and would like asterisk to send voicemail notification messages via exchange. I have had a look at the voicemail.conf file, but I can't see how I would go about configuring it to use an account set up on exchange ? The exchange account would have both POP3 and IMAP access, so how can I tell Asterisk to use the exchange account rather then sendmail ?

...ent. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the register message, at this point, i'm quite stuck. i also noticed that the nonce field is randdata? compared to iptel.org's ser and vovida's vocal. i notice a lot of difference on the sip messages composition. i'm running 0.4.0. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030708/969f2ff3/attachment.htm

Hello! Some time back, I added the Speex protocol to my version of VOCAL (www.vovida.org, VOIP tool). Recently, I also added PESQ (automated voice quality testing algorithm) to my tool and have been running some tests on a clean network. The source file is a woman reading some phrases meant to test various aspects of codecs... Speex has a respectable result of 3.67 Some other c...

Hi friends ! I want to add stun functionality in asterisk. can anybody give me some hint that how can i start that. thanks in advance Deepak Dhiman

...dband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear the other party (to be more precise I can hear first two secs then nothing). So it must be the incoming RTP is blocked on Linksys. Here I think STUN server enters the game and give some help? I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touch...

Asterisk will become a real ip tel softswitch or is going to other way ? like vovida .... regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031024/361291f9/attachment.htm

Hi- In trying to track down a possible memory leak in asterisk, I've discovered that the "show memory allocations" command crashes asterisk (causes it to stop handling calls, although it doesn't seg fault). The related "show memory summary" works however. Before I post this to the bugs list, can someone else please confirm this problem? You need to enable malloc

Hello all, It's my try to make some 'emulation' of vovida's b2bua using asterisk. I was in rush while writing it, so I sure there is much code that can be cleaned, great that not too much. :) http://dslmax.boom.ru/asterisk_b2bua_v0.1.zip cdrradius and agi script inside. __ Mike Tkachuk

Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 "No proxies are up - can not send message to anyone" Xlite is not able to register to the asterisk server. Is there anyth...

...s 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 address NAT is to use STUN or uPNP.. So my question is would it not be better to couple STUND (Vovida.org) with Asterisk and then use nat=yes in the sip.conf for UA's that do not support STUN, instead of using SER which would be like learning Asterisk all over again and would require you to learn how to use the SER config language to manage your NAT transtaltions.. To me the idea of using...

This is an idea from FreeSWAN, which was implemented in the recently released version 1.0. Basically the idea is that FreeSWAN sites automatically encrypt traffic between them when possible, without having to set up the link ahead of time. How this works is: The sites publish some info in DNS. FreeSWAN gets some traffic destined for that site. - looks up the info in DNS - if the info is there:

...O this would work, but I currently like my unlimited plan with Vonage. Would anyone like to enlighten me? I have done numerous searches and I've included a few postings that were mostly not answered. http://lists.digium.com/pipermail/asterisk-users/2004-May/046855.html http://www.vovida.org/pipermail/mgcp/2001-May/000571.html Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040728/30de3b01/attachment.htm

Hi All, I have an equipment loaded with 4 X100P (numbered 1-4)) and one T400P (numbered 5-8). Everything works fine except that I cannot use one of the FXS ports (number 5). If I configure zapata.conf to recognize it, the whole system voice quality suffers. I've tried already to switch PCI slots, with no results. Below is a snapshot of my /proc/interrupts, maybe this can shed some light on