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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 i...

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

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'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

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

...#39;en') -- 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>

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

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'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

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

...erisk PBX 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 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

...;Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Second edition of my * book has been released The second edition of my Asterisk book "VoIP Telephony with Asterisk" is now in print. It's reorganized and expanded. TKS Paul Mahler Paul Mahler pmahler@signate.com www.signate.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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

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

I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one

...Second edition of my * book has been released >> >>The second edition of my Asterisk book "VoIP Telephony with Asterisk" is >>now in print. It's reorganized and expanded. >> >>TKS >> >>Paul Mahler >> >> >>Paul Mahler >>pmahler@signate.com <mailto:pmahler@signate.com> >> >>www.signate.com <http://www.signate.com> >> >> >>_______________________________________________ >>--Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- >> >>Ast...

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

...t trade price twice over. Just pulling your leg. No hard feelings :-) Ad. PS I suggest a working title of "Small Biz Robust Asterisk Solution - SBRAS" On 18 Jan 2004, at 6:00 pm, asterisk-users-request@lists.digium.com wrote: > Message: 6 > From: "Paul Mahler" <pmahler@signate.com> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] ultra-cheap asterisk box - no such thing > Date: Sun, 18 Jan 2004 09:00:08 -0800 > Organization: Signate, LLC > Reply-To: asterisk-users@lists.digium.com > > My Dell 400sc server was $...

...To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com >Subject: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout >question > > >>Signate sells a single server that can get you to the call volumes you >need. >> >>Paul Mahler >><mailto:pmahler@signate.com>pmahler@signate.com >>www.signate.com >> >[snip] > >Past conversations on this topic have generated quite a bit of >controversy within the Asterisk development community, both publicly >here on the list forums as well as in quite a few more quiet >discus...