[asterisk-users] Asterisk SIP and SRTP
Hello, are there any plans in including SRTP into Asterisk? The patches in http://bugs.digium.com/view.php?id=5413 are pretty old and do not work with asterisk 1.6.0. Thanks, Artem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6
Hi, On Tue, Sep 23, 2008 at 01:05:17PM -0600, Joseph wrote: > I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage > but I think this version has a problem with RFC2833 DTMF signaling and I > don't think there > will be any newer version available anytime soon on portage. > > I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys > and Sipura); should I go to 1.6 or 1.4? You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in the overlay. # emerge layman # layman -a voip Regards, Artem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK supported Video phone
Hi, Gordon Henderson schrieb: > On Thu, 4 Sep 2008, Tharanga wrote: > >> Hi folks, >> >> Can some one recommend a good video phone for asterisk (SIP.IAX2). I need >> better quality, duarability. and should support various video codec's >> .(Codec auto negotiation support id prefferble) > [...] > > Some people have reported good results with the BT Videophone 1000 units > too.. (avalable for <£60 a pair, but they need to have the early s/w > release on them) Are the BT Videophones full featured SIP phones, or do they have to be "cracked" to work with asterisk? Do you have some more info about these phones? Thank you, Artem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with 2 Asterisk servers on same LAN
Hi, On Sat, Sep 06, 2008 at 09:52:45PM -0400, hugolivude wrote: > OS = CentOS 5 > Asterisk = 1.4.21 > Router = WhiteRussian 0.9 > > Not sure whether I have a problem w/ Asterisk or White Russian config, > so I'm posting to both lists. > > I _think_ I have the ports forwarded correctly on my router. I set > DESTINATION ports for the SIP & RTP ports above such that ports 5060 & > 1-2 go to 192.168.2.160 while ports 5070 & 21000-25000 got to > 192.168.2.170. Frankly I find the Firewall GUI a little unintuitive – > here's what /etc/config/firewall looks like: > > forward:proto=udp dport=5060:192.168.2.160 > forward:proto=udp dport=1-2:192.168.2.160 > > forward:proto=udp dport=5070:192.168.2.170 > forward:proto=udp dport=20001-25000:192.168.2.170 > [...] I think, that your problem is the port forwarding on the WhiteRussian box. Please try to setup the port forwarding in /etc/firewall.user instead of /etc/config/firewall. /etc/config/firewall has never worked for me. Try something like this: iptables -t nat -A prerouting_wan -p udp --dport 5060 -j DNAT --to 192.168.2.160 iptables-A forwarding_wan -p udp --dport 5060 -d 192.168.2.160 -j ACCEPT iptables -t nat -A prerouting_wan -p udp --dport 1:2 -j DNAT --to 192.168.2.160 iptables-A forwarding_wan -p udp --dport 1:2 -d 192.168.2.160 -j ACCEPT Regards, Artem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Combine sip audio and video from different sources
Hello list, Is it possibe to "multiplex" audio and video from 2 different sip phones and transmit both during one single call? I would like to setup something similar to the Cisco Video Advantage. Lets asume the following setup: Both parties have a sip hardphone and a sip video softphone. Both, the hardphone and the softphone register them self to asterisk. If one party calls the other party then the hardphone should ring. If the other side picks up the phone, then video should be transmitted between the soft-phones and audio between the hard-phones. (I think this can be achieved with some dailplan rules, setting up the softphone to auto-pickup and disabeling the audio codecs on the video phone (we just establish 2 seperate calls one for video and one for audio)). But the more difficult thing is what to do if one side has just a sip-video phone without a stand alone hardphone. Now audio and video from the hard and softphone must be "multiplexed" and transmitted to the single sip-video-softphone, so the other end can talk and watch using just the soft phone. Is something like this possible? Any ideas? Thanks, Artem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
