similar to: SIP error 407 - can't make outgoing calls

But for us? _____ From: William Boehlke [mailto:william.boehlke@signate.com] Sent: Wednesday, January 11, 2006 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Second edition of my * book has been released $39.95 retail. _____ From: asterisk-users-bounces@lists.digium.com

i'm getting a tick every second or so on all my calls. All channels are zap channels. Does anyone know how to fix this? Thanks! Paul Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

Signate has an immediate opening for a qa/release engineer for our line of VoIP telephony products. Release Engineer Signate is rapidly growing and profitable. We are about to launch a new line of telephone software products. That?s where you can come into the picture. You would support Signate's software development team by reviewing new and changed code, tracking and auditing change

I would like to be able to detect when someone dials *. What I'd like to be able to do is exten => *,1,Answer and catch it when the caller pressed the * key. Thanks! Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

I need to dial an extension that tells me what extension I'm dialing from. I'm running a bunch of analog phones off a channel bank to * over a T1. I have the following in extensions.conf. exten => 98,1,SayDigits(${EXTEN}) This says the digits the caller enters on the keypad, not the extension they are calling from. Thanks Guys!!!!!!!! Paul Paul Mahler pmahler@signate.com

Might be a stupid question, but is there a default gateway set on the 7960? -----Original Message----- From: Paul Mahler [mailto:pmahler@signate.com] Sent: Thursday, December 18, 2003 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT? I have a 7960 running behind a firewall running NAT. From a telnet session to the 7960, I can't ping

How does it compare with the O'Rielly book? Does it include information on CVS, or primarily on stable? Can it be provided to customers, or is it more sysadmin oriented? Regards, Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Mahler Sent: Thursday, January 05, 2006 9:45 AM To: 'Asterisk

Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten => 99,1,VoicemailMain(${EXTEN}@inside) exten => 99,2,Hangup Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

I am using the following macro to dial a ZAP channel. When I dial in, * answers and I go to voicemail. I never hear any ringing, though. It doesn't work with the Ringing command before or after the Dial command. [macro-zapdial] ; ; call a ZAP extension for ${ARG2} seconds, and then voice mail ; ${ARG1} - Extension ; ${ARG2} - Time to ring exten => s,1,Dial(ZAP/${ARG1},${ARG2}) exten

As you all know, one of the biggest criticisms of Asterisk has been the lack of documentation. Paul Mahler of Signate has taken the initiative and is writing an introductory guide to Asterisk that Digium plans to help publish. This is a guide for beginners, not for gurus. I would like to see the community help Paul in his efforts to help document Asterisk and make it easier for someone to start

I agree. I haven't had a problem using CAT-5, even for long runs, however it's not a real T-Carrier cable and I didn't know how old the PBX is. Paul >I have not in my experience seen any problems with using a Good Quality >Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you >should be fine. As far as the shielding goes, I use UTP cables and >Connectors

I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 => 3213,Bill Smith Thanks! Paul Mahler mail:pmahler@signate.com phone: 650.207.9855 fax: 877.408.0105 -------------- next part -------------- An HTML attachment was

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm

Why is using ulaw or alaw an unlikely scenario? I wouldn't use anything but ulaw\alaw. The Bells can compete on price and will if they have to. Where they CAN'T compete is quality. If there were something better than 711, I'd offer that. Well, there is 722, but not many things support it. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original

I am trying to make outbound calls from my Asterisk client through a remote Asterisk server with IAX. In iax.conf on both sides [dar] context=trusted secret=xxxxxx type=friend host=192.168.1.1 in extensions.conf at the client making the call Exten=_1NXXNXXXXXX,1,Dial(IAX2/dar:xxxxxx@192.168.1.1/) What goes in extensions.conf at the remote server? What is needed for the

I have a strange problem rolling through my 7960 phones. One or more of the phones goes crazy when the first digit is dialed. The display flashes repeatedly, the phone does a bunch of stuff, sometimes it even reboots. It's not the powered switch, the same thing happens with a different unpowered switch. It's not the phone, the problem moves from phone to phone. If it's

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, May 25, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or

I don't think they are doing it with one Asterisk box. They did say "one rack of servers". Well, that might mean up to 50 computers if they are using blade servers. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Todd Sent: Thursday, February 02, 2006 10:21 PM To:

Here's our recent announcement of our new Asterisk Installation CD set: Signate has announced its new Asterisk Installation 2005 CD Set. It's, a complete software PBX (private branch exchange) telephony appliance in a single package. The CD set installs Linux pre-configured for telephony, a stable 1.0x distribution of the open source Asterisk PBX, and Signate's optional, free PBX

Need new Music on Hold for your PBX? Signate is happy to make a variety of classical music selections available, sampled at rates that are appropriate for telephony. There is no charge. The selections feature Elena Kuschnerova, pianist, and Lev Guelbard, violinist, playing public domain pieces that will give callers a classic impression of you or your company . Click on the link to see a list