similar to: Why don't I get a ringing 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

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'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

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

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

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:

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

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

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

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

If anyone is interested in getting a book on asterisk I would recommend checking out http://www.saww.net/asterisk/

Hello We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work with our call agent. Unfortunately **VERY** frequently, asterisk stops responding and goes to 99.9% CPU. There is no debug output or other information that indicates there is a problem... Rather than continually restarting, can anyone make suggestions as to how we can track this down **OR** has anyone got the

I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08

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 =>

Got my latest Linux magazine (www.linux-magazine.com) and fetured is asterisk in home network. I've also been in contact with Novel/SUSE about their asterisk pakages. *Reinhard Max the maintainer. He has hinted at new packages for SUSE 10. The current ones work well (in production) however he is unsure about the new zaptel intergration.... but I'm keeping my fingers crossed! * -- A.G.

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