search for: viageni

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2013 Apr 23
0
Asterisk 11.4.0-rc1 refuses to use the TURN server
...s on both the Asterisk side and the Chrome side, we've decided to just simply go with a TURN relay for RTP packets until the issues are resolved. I configured rtp.conf so that all of the STUN related entries are commented out, and I use the following TURN configuration instead: turnaddr=numb.viagenie.ca:3478 ; ; Username used to authenticate with TURN relay server. turnusername=myusername%40gmail.com ; ; Password used to authenticate with TURN relay server. turnpassword=p at ssw0rd I also use the same configuration on the client side. When running a tcpdump, I see that there is traffic to/f...
2007 Aug 29
5
Ringing sound doesn't work
...ave these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension if I use this one. I just get silence until someone answers. How come? I use Asterisk 1.4.10. I have attached my extensions.conf file to this email. Thanks, Simon -------------- next part -------------- [globals]...
2008 Sep 15
0
rc6: Dunno what to do with STUN message 0101 ??
Having some trouble with sip behind a nat. So tried: stunaddr = numb.viagenie.ca in sip.conf. Didn't help so tried stun debug: asterisk*CLI> stun set debug on STUN Debugging Enabled STUN Packet, msg Binding Response (0101), length: 36 Found STUN Attribute Mapped Address (0001), length 8 Ignoring STUN attribute Mapped Address (0001), length 8 Found STUN Attribute C...
2010 Jul 15
0
Last call for AstriCon talks
...tutorials, and case studies. We've got some really great talks from a fantastic array of speakers. While I don't want to publish the whole list just yet (that'll be next week) I can tell you that there are how-to talks on IPv6 (a double-session!) by the developers of the code (Viagenie), VoIP encryption techniques by the developer of some of the code (Terry Wilson), and a practical session on SIP security by the author of SIPVicious (Sandro Gauci.) If you've talked with me about giving a session, but not actually put it into the then it's not on the consideration...
2012 Apr 01
0
10.3.0: gtalk_request: No XMPP client to talk to, us (partial JID)
..."gtalk/andy-gtalk/+1xxxyyyzzzz at voice.google.com") in new stack [Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP client to talk to, us (partial JID) : andy-gtalk gtalk.conf [general] context=google-in ; Context to dump call into allowguest=yes stunaddr = numb.viagenie.ca bindaddr=0.0.0.0 externip=aa.bb.cc.dd disallow=all allow=ulaw [andy-gtalk] username=<username>@gmail.com context=google-in connection=andy-jabber gtalk show settings Global Settings: ---------------- UDP Bindaddress: 0.0.0.0 Stun Address: 66.228.45.110...
2008 Oct 16
0
Sharing my Asterisk + SPA3102/PAP2 setup: What I've learned in 1 week.
...Dogs know nothing about telephones. I have: * studied a lot. DO THAT. if you find an error, you will not find the cause if you do not know where to look, where to change something, where to disable something. * Someone ponted me to this document, which I started with. Nice to start. http://www.viagenie.ca/publications/2007-03-apricot-asterisk-primer-blanchet.pdf * Fedora 9 (my desktop and the same at home). IP: 192.168.1.141. IPTables disabled. * installed asterisk as a fedora root user: # yum install asterisk asterisk-sounds asterisk-voicemail I bought: * 1 Linksys SPA3102 (1 FXS, 1 FXO) *...