Displaying 6 results from an estimated 6 matches for "viagenie".
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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/fr...
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
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[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 Ch...
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 l...
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
E...
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)
*...