similar to: Terrible TICKING sound

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 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

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

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

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

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

Does anyone know the best approach to take for handling 100 analog phones? It seems to me that a chassis like Carrier Access or Adtran would work. The chassis would do much of the hard work of converting the analog sound to data. Any recommendations on hardware for the chassis? ...Jeff

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 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:

Paul, I wholly agree with what you're saying - I too ensure that we have at the very minimum, a set of full spares. However, this thread really has the wrong name at this point... We're now looking at embedded solutions, in the same way Cisco has with it's ICS 7750 solution. I'm looking to build a robust embedded solution, that we can run in tandem - of course, we want to

I am trying to find a manual for the Carrier Access Adit 600. Does anyone know where I might be able to find one? Thanks -Jon -- Jon J. Brandon jon@monsoonretail.com http://www.monsoonretail.com

I'm using the stable branch. Is voicemail or voicemail2 deprecated? TKS Paul pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040502/1b0ab572/attachment.htm

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

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

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com

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 am using a IPDialog siptone II. I can take incoming calls, but when I try and make an outgoing call I get a SIP 407 error. Can some kind soul explain to me what I am doing wrong? Here's what I found in the wiki: If a proxy does not accept the credentials sent with a request, it SHOULD return a 407 (Proxy Authentication Required). The response MUST include a Proxy-Authenticate header

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

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