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After getting my feet wet with asterisk@home, I want to set up a second asterisk box to add a call shop billing and other add-ons such as LCR. My question is as follows. Is there a backup program that will save to a tape drive or a USB CD Writer so if I mess up an install I don't have to go through a complete reinstall? I saw a few programs out there but they required X windows and from

I am running Asterisk@Home Voicemail works fine but does not email out the voicemail attachments. Any suggestion? ----------------------------------- Voicemail.conf [general] #include vm_general.inc #include vm_email.inc [default] 201 => {password},Jeff G Laptop,jrglass@columbus.rr.com,,attach=yes --------------------------------------------------------------------- Sip.Conf [201]

Just installed V 0.4 of asterisk@home Programmed up 3 sip budgetone extensions, they call call each other fine. Tried to dial '9' for an outside line through an X100P to a packet8 ATA but got 'all circuits are busy now'. Here is the console output. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-8d25' -- Executing

It said 'include zapata-channels.conf', where this line wasn't commented bij the ';'... Could you post me a working example of such a config (or a part of it, for the X100P cards...? Thanks guys! Message: 9 Date: Sat, 19 Mar 2005 18:04:26 -0500 From: "Jeff Glassman" <jrglass@columbus.rr.com> Subject: [Asterisk-Users] newbie question To: <asterisk-users@lists.digium.com> Message-ID: <000501c52cd8$02c10750$0200a8c0@newgems> Content-Type: text/plain; charset="us-ascii" bram kortleven Wrote >"Message: 6 >Date: S...

I have asterisk with two zap channels which are analog ports off a T1. They each have a inward DID number If they are used for outgoing they show the T1 main number not the DID's number. Is there any way to send caller ID of the inward DID number not the main number Jeff

Hi All, The past day or so I've setup a new * server based upon the Xorcom Rapid ISO. It did as promised; wiped the base system, installed Debian OS, installed Asterisk with a dummy configuration. So far so good. If I could get the config from my existing * server migrated to the Xorcom box the I'd be ready to roll. Essentially, it would be the same as I have now on Fedora Core 1, but

Hello, I am trying to configure the my asterisk box here with the following **iax.conf*** [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx ***extensions.conf:*** exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx@NuFone/${EXTEN} exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan.

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

I have a PCPHONELINE SIS phone set it up to asterisk Registered SIP '205' at 24.172.221.22 port 2770 expires 120 (Port changes every time) Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 24.172.221.22(24.172.221.22 is my phones IP) Anyone have an idea what the problem is? Jeff

I has Asterisk up and running on my IP address. I put a Linksys router in front of it and forward the following ports 22 TCP 5060 UDP 10000-20000 UDP 80 Both None of my x ten phones work. They register but I get an message Authentication Required [202] username=202 type=friend secret=****** qualify=no port=5060 nat=yes mailbox=202 host=dynamic dtmfmode=rfc2833 context=from-internal

My DID with Telasip is disconnected and my Asterisk box won't register with them. Anyone else having problems with them? Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050814/886ceb38/attachment.htm

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf