search for: mahler

Hi all, I'm currently gearing up for a possible PBX replacement project using Asterisk, and I'm just breaching the iceberg of information that's available. I typically like to have something thick with pages in front of me. Mahler's book was the first one to come up and it seems like a good place to start. However, the big name bookstores tell me it'll take up to three weeks, and this project simply can't endure that wait. Does anyone know where it's possible to get a paper copy *quickly*? #2, I'm plan...

Anybody know of a UK supplier of "Voip Telephony with Asterisk" " by Paul Mahler ? I've searched on the web, and the only suppliers I can find are US based, and the postal charge is as much as the book. cheers -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856...

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

Hi All, I am not sure if this is the same thing as the hang on exit bug, so sorry if this is a duplication of previous stuff. Essetntially I am experiencing ssh hangs with about .5% - 1% of my connections. I am running 2.9p2, on Solaris 7. I actually have empirical data on the hangings, as I wrote a script to create these connections in an endless loop, setting an alarm so I could recover

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 s...

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

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

...9;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 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 reorganiz...

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 have the following entry in zapata.conf, but I don't get stutter dialtone when there is a message waiting. Suggestions? Please? callgroup=1 pickupgroup=1 callerid="Paul mahler" <100> context=inside mailbox=100 channel => 1 Thanks, Paul

...TY FOR ANY GENERAL, PUNITIVE, SPECIAL, DIRECT, INDIRECT, CONSEQUENTIAL OR INCIDENTAL DAMAGES, OR LOST PROFITS OR ANY OTHER DAMAGES, COSTS OR LOSSES ARISING OUT OF YOUR USE OF THE FREE MUSIC ON HOLD FILES, EVEN IF SIGNATE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES, COSTS OR LOSSES. Paul Mahler www.signate.com

...nguage '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>

There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

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 know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't

...39;s book is just right for rounding out the edges when getting started. I managed to temporarily migrate our T-1 Asterisk system to a Analog asterisk system on information in Paul's book alone. Nicely done and a neat bit of help in a pinch. Just my $0.02 USD you understand. :) RandyW Paul Mahler wrote: >Hi Greg, > >My book is a good place for a beginner to get started. I also find it to be >useful as a reference for Asterisk. It's not an advanced book, there are >advanced features it doesn't cover, for example AGI or the management >interface. > >It shoul...

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

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

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

...all kinds of abandoned ssh processes lying around that have to be manually killed. Does anyone know if there is going to be a fix for this problem or how to make it work correctly? Here's an example of what I see when I run a remote xterm, and then close it immediately: polycut:~> ssh -n mahler xterm <I immediately hit CTRL-d on the xterm and the the following error in the original shell> channel_lookup: 0: bad id: channel free client_input_channel_req: channel 0: unknown channel <I hit CTRL-c to get control of the original shell> ^CKilled by signal 2. polycut:~> Thank...