[Asterisk-Users] app_conference, CVS HEAD, SIP and Xen
I have Asterisk running in Xen virtual machines. Unfortunately, this kind of virtualization makes a real time clock impossible, which in turn makes ztdummy or a Zaptel driver impossible to load, which also makes MeetMe conferences impossible. As an alternative, I have downloaded, patched, compiled and installed the app_conference source code against the headers in Asterisk CVS HEAD. I can load the module into Asterisk and even connect to a conference channel with two phones. But each phone cannot send audio to each other. They just connect and go silent. Is this the current state of app_conferences development? I have read few comments online that sound like some people are using app_conference with sound. -- Lee Azzarello Network Engineer Progressive Solutions +1 212 937 8939 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: app_conference and AGI
The README in the source code states: "app_conference doesn't have DTMF-activated features or anything like that." I'm curious how you got audio working on your compliation. I am running CVS HEAD + app_conference in a Xen virtual machine. I can connect to the channel but there is no audio. Here are my configs and Asterisk's output: http://lee.97montrose.org/hacking/app_conference.txt -- Lee Azzarello Network Engineer Progressive Solutions +1 212 937 8939 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
>> > The fact is, there is not ONE sip or iax softphone that is as easy to > use as skype for the average user. The sad thing is it doesn't have > to be that way. Spend the $100 and get her a IAXy that's pre configured to your local Asterisk server. Then she can use an analog phone to call you for free. -lee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterix-users] CISCO 7910
On Sun, 2005-03-27 at 14:18 +, Titoy LeBoss wrote: > Hi everybody !!! I'm a french student working on an Asterisk's projet . I > think it's very easy for you : > configure 2 CISCO 7910 and Asterisk in order to make call between both. > (Scuse for my poor english ;) ) > I have install Debian and Asterisk thanks to the CD Xorcom Rapid. > I have set-up DHCP and TFTP servers for the IP phones. > On the tftp, i have put the *.bin files for my phone, SIPDefault.cnf and > OS79xx.txt > Can you give me an example of SIPAdrMac.cnf ??? > Can you explain me what i have to modify or add on Asterisk?? > Skinny.conf ??? SIP.conf ??? > And have i to add Channels in CLI Interface of Asterisk ? You have to consult the Cisco documentation for the phone's firmware. On the asterisk side, you just have to edit sip.conf to let the server know about the phone and it should register. If you type 'sip show peers' you should see the phone the server knows about. -lee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small qos switch
On Sun, 2005-03-27 at 21:19 -0500, Moody wrote: > I don't want to move this thread towards a discussion of Sveasoft, but > I would ask anyone considering this option to make sure they do some > reading about Sveasoft and their version of "opensource" before > sending them a check. > > IMHO, Do the community a favor and check out OpenWRT (see earlier > posting in this thread) or one of the other options instead of > supporting Sveasoft. VPN functionality is a big deal and the OpenVPN support OpenWRT speaks of may not follow the IPSec spec since it's not listed on the VPN conformance test: http://www.vpnc.org/testing.html The Sveasoft has multiple firmwares and only one can be loaded at a given time so that's another limitation. So there's no promise it'll interopperate or do everything you want. On the open source conformance, Sveasoft addresses this directly on their own forums: http://www.sveasoft.com/modules/phpBB2/viewtopic.php?t=2823 If it works...it works. Take it for what it is. -lee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
