As Kevin Fleming Says "So Long, And Thanks For All The Fish!", We Say Thank You - And Look To The FutureReport
It's amazing what you can learn in a few days...
Having just found out that Queen Elizabeth has a great sense of humor, it has now emerged that Kevin Fleming - a man who (both with and without his moustache) has been an amazing contributor and influencer in the Asterisk project is set to move on to a new challenge outside the project - but still within the realms of Open Source.
Kevin has been involved with Asterisk for 7+ years, and has been both a thought leader and a powerful voice in the Asterisk world during that time. I…
I've been with Digium for just over seven years, and it's been an incredible experience that I wouldn't have traded for anything. When Mark Spencer invited me to visit Digium (and Huntsville) in early 2005, I could not have dreamed that I'd end up working for such an exciting, innovative company, finding a wife, and meeting hundreds of people (many of whom are now friends) around the world. It's been a time of tremendous personal and career growth, and my wonderful colleagues at Digium and in the Asterisk open source community have been directly responsible for most of that. Recently, though,…
Really it is miserable.
I bring 8 Digium Phone D40 and I used them with a customer, the voice quality is bad internally (between the extension), there is no clearance at all ! We are hearing the voice like another person.
The used codec is ulaw.
The firmware version is: 1_1_0_0_48178
Even at the web based configuration at the phone it self, I am not able to do reboot (there is no reboot button) and I can do this only from the Phone it self.
From the speaker, the voice is very bad and weak.
I am really feel disappointed why I did not use…
I have two setups with SIP hardware phones as extensions and POTS lines as trunks. Internal SIP to SIP calls are crystal clear, but all calls bridged to POTS have a significant amount of static noise. The problem is that if I plug a POTS phone directly into the line, there is almost no static noise - the line is clean. It's like Asterisk (or the hardware) amplifies the static noise. What I've tried so far:
1. Connect Asterisk with a short cable directly into the master phone socket, where it enters the building. 2. One of the lines carries ADSL…
i have strange problem with AGI (asterisk 220.127.116.11) when i use Dial from dialplan everything is ok when i dial from AGI script there is missing SIP Status 180 ringing and 183 session progress
DIAL without AGI
196.356479 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE sip:firstname.lastname@example.org, with session description 196.356768 10.0.0.213 -> 10.0.0.193 SIP Status: 401 Unauthorized 196.365709 10.0.0.193 -> 10.0.0.213 SIP Request: ACK sip:email@example.com 196.370028 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE sip:firstname.lastname@example.org, with session description 196.370503 10.0.0.213 -> 10.0.0.193 SIP Status: 100 Trying 199.797325 10.0.0.213 -> 10.0.0.193 SIP Status: 180 Ringing 199.797932 10.0.0.213 -> 10.0.0.193 SIP/SDP Status: 183 Session Progress, with…