Greetings- Working with the T.38 gateway functionality that is sparsely documented  , Im attempting to get the following functional: Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box in question) -> SIP Provider The problem ..
Greetings- As many of your are Polycom experienced, I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), Im finding an instance where, using intercom/paging functional..
Greetings-I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 18.104.22.168 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=..
Greetings-I have an odd scenario where I need to dial an extension (lets call it 555), the system prompts for a list of voicemail boxes, then once complete, allows the caller to leave a voicemail that is sent to all voicemail boxes previously specified…
Is anyone aware of a way to replicate parts of the AstDB to another Asterisk install?For example, to export all CF entries on a FreePBX based system to another system running FreePBX, I might do:asterisk -rx database show | grep CFThis gives me a l..
Greetings-Im running some USB DAHDI hardware on a system with a tickless kernel. The audio quality is quite poor. Could the tickless kernel be to blame? If so, when recompiling a kernel that is *not* tickless, is there a recommended KERNEL_HZ val..
Greetings-Ive got a curious project that I could use some input on. Id like to use Asterisk to record some audio channels via USB soundcard. When audio passes through the soundcard, Asterisk should grab that audio channel (CONSOLE?), and write it..
I have a site with Polycom handsets on all the desks, mostly IP650s, some IP550s, and some IP450s as well.I need to update the firmware on the IP450s. However, the firmware simply wont load.The latest firmware (4.0.3 Rev F) supports all phones at t..
Greetings-Im running into an issue as follows, in simplified form:A remote Asterisk box, when registered/peered via SIP to a central server, and makes a call to that central server, is *sometimes* authenticated and calls go through properly (via from-inter..