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I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

I'm having a very strange problem I've been fighting with all day. It's 2:30am, and I'm stuck. I think the problem may lie with one of my SIP providers, but I'm not sure. I have two ways to call into my test Grandstream. I can call a PSTN 360 area code number that will forward to my FWD number, which in turn is registered with my * box on extension 2030. If I call the 360

help me test load a box! I have a new box with four PRIs on a TE405P I will terminate US Toll-free traffic (1-800, 888, 877, 866) for free via SIP to anyone who wants to test. Just email me at matthew@txlink.net if you would, to let me know that you're testing, and with any comments about quality, etc. I have ulaw, alaw, and GSM codecs enabled. To use, just send your call via SIP to

I'm having a problem with one of our newly built call processors. I'm using the stable branch of the asterisk code with a single TE405P on a P4 3.2ghz running kernel 2.6.6 with Hyperthreading and APIC enabled. The machine has 1GB of RAM and HIMEM is enabled in the kernel. Calls terminate properly and everything seems to work okay, but the system died after only being up one day with the

Good evening. I just wanted to take a minute and review my experiences with some of the SIP devices out there on the market. I hope this post will help newbies or someone considering a certain device. I would appreciate any other input on either the devices I am "reviewing" or other devices that I didn't! These devices are deployed in our primary line and small PBX replacement

In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID "1000" and line 2 as SIP ID "2000". Basically I have this set up so that 1000 and 2000 are "lines in hunting" on incoming extension "555". I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here