On 06/05/2012 12:39 AM, Klaverstyn, David C wrote: > Guys, > > All my installs are based on PRI ISDN. I now have a site that I need to > install BRI. As I have not done a BRI install before I’m wanting to get > some information from the people in ..
Anyone near McBee SC that can visit a client location and do a site survey ..
Im wondering if someone has already done a web application that queries ExtensionStatus events. On my web site I have an extension listing. Next to each number Id like to add an icon or something that shows the extension status. Id like this status..
We found this morning we had no SIP connection to another site. sip show registry on the main site gave no authentication. sip show peers on the other site showed the peer unspecified. The odd part about this:doing sip reload on the main site made..
all, This post is in case someone else has this problem. The cause of the issue turned out to be one of the site technicians having the same extension registering from his laptop as the ATA we were testing. His laptop wasnt always connected to the vo..
> I have a site that moved to the latest 1.8 revision, and began to > have problems with phones in far away places (South America, > and the MidEast). > > What I see is that when a Dial() is issued, the sip channel driver > sends out an INVITE to ..
I have a site that moved to the latest 1.8 revision, and began to have problems with phones in far away places (South America, and the MidEast). What I see is that when a Dial() is issued, the sip channel driver sends out an INVITE to the phone.V..
Dear Chris How about to use 2 Asterisk system interconnected through Wireless solution point to point one system should be for ISDN BRI gateway with Digium PCI card and the other server for extension voice mal and so on. for covering distances it w..
We have an old analog phone system running Asterisk 1.2.13 (not my choice lol). Everything has been working wonderfully until today. The site is experiencing dropped and missed calls. When I tried calling the site, I did get through however the CLI ..
folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations t..