search for: signate

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 optiona...

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

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...

...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 <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

We still have a seat open in our Asterisk training course next week in London. You can find more information at our Web site, www.signate.com I'm going to be teaching the class. Paul

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

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 anything outside the subnet the 7960 is on, th...

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 scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040317/f0b03f5b/attachment.htm

...of handling 2500 SIP-SIP calls with no transcoding, including RTP sessions, if on an operating system and hardware that is appropriately configured. This puts to rest some who have claimed that 5000 channels is "impossible" with Asterisk regardless of platform, at least according to Signate. b) It is unclear if other channel drivers (IAX and Zaptel, specifically) have had any testing with significant numbers of channels. c) It is unclear if anything other than pure RTP passthrough is viable in these configurations. Maybe IVR causes collapse. ? 2) Still no claims or co...

...ription_ We are looking for an expert LAMP (Linux, Apache, MySQL, Perl, and PHP) developer with some Asterisk experience who is based in Western or Eastern Europe or Asia. We can work with an individual or an organization. You must be fluent in English. We need you to help expand development of Signate's core software products. As part of the Signate development team, you will design, develop, debug and test code for new and existing systems. You will work independently. You will be given a great deal of freedom to exercise your knowledge and experience. In return, we will expect high qu...

Greetings. Please reference: http://lists.samba.org/archive/samba-ntdom/2001-September/019783.html That list is listed as "inactive" on the samba site, so I'm following up here. Does anyone know if this functionality was ever achieved? We just replaced out NT4 PDC with samba and I'm facing this exact problem. I do not relish the thought of having to "touch"

...39;) -- Playing 'vm-intro' (language 'en') == Spawn extension (macro-zapdial, s, 3) exited non-zero on 'Zap/49-1' in macro 'zapdial' == Spawn extension (main, 100, 1) exited non-zero on 'Zap/49-1' -- Hungup 'Zap/49-1' Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com>

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

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...

The second edition of my book "VoIP Telephony with Asterisk" is now in print and available. You can find out more about it at our web site http://www.signate.com/products.php This book is written for beginners. It will make it easier for you to get started. The second edition is reorganized and expanded. Thanks, Paul

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

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

...X Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:503@216.38.136.149> Proxy-Authenticate: Digest realm="asterisk", nonce="230958ab" Content-Length: 0 The password at the phone is the same as the password in sip.conf. Thanks! Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training

Is there anyon here currently in New Zealand that use asterisk, I need to help getting voice and internet services. I will be moving in a week. Any help would be great. Please use the details below to get ahold of me. Thanks in advance. James Jones Signate, LLC james.jones@signate.com 415.442.4012 (office) 413.771.1402 (office) 413.977.6482 (mobile) 413.667.3105 (fax) 665 Third Street Suite 100 San Francisco, CA 94107-190 Asterisk Services and Training -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.di...

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 [mailto:asterisk-users...