search for: panterafreak

Ok I have a box with 2 nics, one "outside" and one "inside" (192.168.8.x) I want to get connections for sip and iax on the outside connection I have a firewall on the box but have opened the followng ports pbx 5060-5070 udp + tcp pbx 4569 udp pbx 5036 tcp pbx 2727 udp pbx 10000-20000 udp pbx 69 udp This allows me to connect with iax but sip fails registration with a 408. I

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the interface it has a place for me to record new IVR menus. I have to dial *77 to begin recording

I solved this issue by updating the 2003 AD Servers to SP1. Regards, Bob Bostwick -----Original Message----- From: Tim P [mailto:panterafreak@gmail.com] Sent: Tuesday, July 26, 2005 10:18 AM To: samba@lists.samba.org Subject: [Samba] NTLMv2 - wrong password with samba? I have samba 3.0.14-5 installed (installed via Fedora Core 4's Yum) I have enabled "client NTLMv2 auth = yes" in smb.conf When I run "ntlm_auth --user...

I installed a new * server and copied the sip.conf and extensions.conf from my existing setup to the new box. I created my outbound trunk with a different broadvoice account and am able to dial out without issue. I am able to dial all extensions but I go straight to voicemail without any ringing, this happens to both local and outside numbers calling my extensions. I have the correct firewall

I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one and answering it and then hanging up results in the line be permanently busy (the phone called is permanently busy until * is rebooted). Any idea where to start with this one? It seems to me that either the SPA2100 is not registering the end of the call or * isn't.

Using the Xten X-Lite client (free) I am able to connect to a local asterisk@home server and when trying to connect to the remote server (a mirror of the local) I am unable to connect. The first server is a local lan, the remote is using microsofts pptp vpn client to connect. Looking at the diagnostics from X-Lite I see this: <begin log> SEND TIME: 341549001 SEND >>

I am trying to create a system as follows: incoming call --> ivr --> sent to dummy extension 1000 --> redirects user to ring group 2000 --> (ring group consists of extension 3000 and 4000) --> no answer --> sends to voicemailbox 1000 I want to do this to be able to have one extension number (and mailbox) for a person but allow them to have multiple extensions transparently

Maybe this marks me as a real newb but where do I set the number of rings that a phone has before it sends it to voicemail? Also for some odd reason when I ring an extension attached to my sipura 2100 ATA it takes it about 12 seconds to start ringing after I dial it (sits there with dead air on the calling phone). Any idea on these, am I missing some simple configuration switch for either?

Problem 1 - Outgoing: I am able to call out of the * box using the analog line attached to the sipura 3000 but when the person being called answers there is no audio from either end. * registers that the call was answered but passes no audio. Problem 2 - Incoming: When calling into the 3000 attached to * it never seems to pickup the line. The phones don't ring on the asterisk side. I used

I have samba 3.0.14-5 installed (installed via Fedora Core 4's Yum) I have enabled "client NTLMv2 auth = yes" in smb.conf When I run "ntlm_auth --username=user --domain=MYDOM" it connects fine (change user and MYDOM to be my user and my domain) When I run "ntlm_auth --username=user --domain=MYDOM --diagnostics" it fails on all tests with "wrong password"