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...;m using asterisk 1.0.6 and I would let media path be connected directly between the phones without going through Asterisk. I have to it with an AtCom320 (with pa168s chip). I just saw and tryied to do what this page http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%2 0clients%20connect%20directly says. Before going on (with sniffer eth traffic between * and two phones) I'd like to known if it can works. Does anyone just did it? Thanks in advance Gio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/a...

...E----- Hash: SHA1 Hi there, i'm in debian sid 3.1 with kernel 2.6.7, * last cvs & chan_capi 0.3.4b; nt1+ with 2 bri in ptmp (http://www.voip-info.org/tiki-index.php?page=DDI) i tried to install avm c4 following step by step http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20connect%20with%20CAPI step 1. i compiled capi 2.0 support in kernel 2.6.7 step 2. modprobe c4 to insmoded kernel module then: "You may need firmware in /usr/lib/isdn/ and a suitably configured /etc/capi.conf " i googled around for a suitable firmware (c4.bin) but the only one i was able to fi...

Hi list, I have some nontrivial questions. I am no telecommunication guru and I will explain it with my simple words. I hope someone can help me with these issues (with Asterisk 1.0.3): - If I call outside (with Nikotel to German Telekom) there is a remote hangup after 2 minutes. I've seen other people posting this but nothing helped. I luckily managed to get around this issue with the

Hi, I am attempting to connect my dell SC420 to a 32" LCD at 1366x768. I have attached my currect config file and currect X start log. How do I massage the xorf file to run at the native resolution of the LCD? It currently runs but is in a virtual mode. The screen scrolls. Any help is appreciated. I am using Centos 4.1. Jerry -------------- next part -------------- An embedded and

Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the

Here is the setup: Phone A (in NYC) on own bandwidth. Phone B (in LA) on own bandwidth. Asterisk box in Houston,TX on own bandwidth. Both phones contact asterisk to register. Not much bandwidth used for this as it is a few packets every hour or so. Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk calls phone B. Both phones are connected and both people are talking.