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Hello all, With some help from people in #asterisk on freenode, I've managed to get incoming SIP calls working. Outgoing calls however are however a different matter. My whole working (incoming calls only) SIPgate configuration can be found here. [1] When I uncommon what's in there, nothing works. There doesn't appear to be any useful error being logged , even when debug is

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules..... and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever.... but the problem is now I can't get past the boot up and the

What happens if you take out the Zaptel I/F's? If it boots, you can correct whatever you did then replace them. hth -----Original Message----- From: Chris Earle (CBL) [mailto:cearle@cbltech.ca] Sent: Tuesday, February 28, 2006 7:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot boot machine up after working on zaptel.... Hi all, hard for me to explain this, but