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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] username=201 type=friend secret={ACCOUT PASSWORD} qualify=no port=5060 nat=yes mailbox=201 host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Jeff G Lapto...

...id '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: Sat, 19 Mar 2005 22...

...nes to access trunks include => outbound-emerg include => outbound-info include => outbound-local include => outbound-tollfree include => outbound-ld include => outbound-international include => outbound-trunks exten => h,1,Macro(hangupcall) exten => _394.,1,SetCallerId,jrglass@columbus.rr.com exten => _394.,2,Dial(IAX2/12355@bellster/${EXTEN:3},60,r) exten => _394.,3,Congestion ; ############################################################################ ; Extension Contexts [ext] ; ############################################################################...

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

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

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

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

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>