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Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk is 1.6.2.1...

Hi All, I'm currently having trouble with dropped phone calls. The following error message is always in the log. This is a Grandstream GXP-2000 Firmware 1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been occurring on other versions also. [Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission 8acaea6dc4c6e9b5 at

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

Hi guys, I'd like to add a LIVE SUPPORT function to my website. Basically I want a client on my desktop that pops up when someone request help BUT doesn't appear or says offline when I'm not available or have logged out of this function. When a person visiting my website has a question they hot the button to cause a text popup chat to occur. Anyone know of an open source

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?