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...ain=0.0 txgain=0.0 group=1 immediate=no context=incoming signalling=fxs_ls channel=1 zaptel.conf fxsls=1 fxoks=2 fxoks=3 loadzone = us defaultzone=us [dialout] exten => _9NXXXXXX,1,StripMSD,1 exten => _NXXXXXX,2,Dial(Zap/1/w${EXTEN}) exten => _NXXXXXX,3,Hangup Thanks Jason Kawakami www.optellabs.com Salt Lake City, UT

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

...etrieval, leave out a pin in voicemail.conf to make it easy [local] Exten=_NXXXXXX,1,Dial(zap/outgoing channel/${EXTEN}) [long_distance] Exten=_1NXXNXXXXXX,1,Dial(zap/outgoing channel/${EXTEN}) Note that I haven't done the first example but the second one is duck soup. Jason Kawakami www.optellabs.com Salt Lake City, UT

...m only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical Sales Open Telephony Labs, LLC 801.527.2284 www.optellabs.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040804/f3afac3b/attachment.htm

BR C.

...144[0-9]* that I thought would match 01144X. and send the call to the UK but the script is matching 01144207108???? With the first route. Can someone smarter than me help with some samples? Please? If I can get one for 1NXXN. and 01144. I should be able to figure the rest out. Jason Kawakami www.optellabs.com Salt Lake City, UT

I need to be able to dial a long distance provider that uses an access number and an auth code. I would like to be able to program this so that the user can dial 8 and then the long distance number, asterisk will hopefully do everything in the middle. The sequence to accessing the provider is on my traditional phone speed dial as: * Dial local access number * Wait 5 seconds * Dial the auth

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

I am new to Asterisk. I am trying to ascertain the hardware setup (and associated cost) I would need. The documentation in the wiki (and elsewhere) is extensive but I am somewhat lost in product model numbers. Hence I need an initial recommandation to work on. 15 incoming lines, 25 employees). Initial scenario is to use * as a plain old PBX. I need voicemail, ability to transfer calls, ... I

...e currently) -you will probably have issues with that many zap cards in one server. Look at the IAXy as an ATA for the fax machines (be careful, faxing over IP can be FUN!)or look at having a couple of machines sitting next to one another that are trunked together. Good Luck! Jason Kawakami www.optellabs.com Salt Lake City, UT

I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple

Hello All, Just looking some comments from gurus about this proprietary systems and phones: Inter-Tel Eclipse2 Model name: IP PhonePlus I did not find anything useful or reasonable about their products on their website or even in Internet.... except sales. -- Thanks and regards, Vasyl Rublyov

...n set * up to work properly. check online for a tool called a digit grabber and connect it to one of the ports on your vm system (assuming that it is currently an analog integration) and make a few calls to see what your phone system is sending your voicemail system. good luck Jason Kawakami www.optellabs.com

Hey all, Did I see something on here about using an AGI script to do reverse lookups via anywho.com? I have a PRI that only gets caller-id number and no Alpha. TIA, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

...hat sends station name across the d-channel without any reverse lookup DB. Now that being said I am no expert on d-channel messaging so I can't really answer the question on how/if we can pass the CALLERIDNAME across a private d-channel connection between * and another PBX. Jason Kawakami www.optellabs.com

Back in the office post-astricon. 1.0.0 running in the lab. YIIIIIIIHAAAAAAAAAA! THIS GUY!!!! rocks. Thanks to Mark for *, Steve and Olle for the conference and to ALL community members. Everyone using * is contributing in one way or another. See y'all next year Jason Kawakami www.optellabs.com

...sics are the same. this is a good one that I used with my first RH install http://members.lycos.co.uk/wipe_out/asterisk/ if you really want to be safe, just do a 'complete' FC2 install and make sure that you check out the README files in the SRC directories. Have fun! Jason Kawakami www.optellabs.com

Hello list, I am new to Asterisk and have a few questions, I hope you can tell me if Asterisk can do what I need. I'd like to create a simple IVR menu that will allow people to enter prices using their telephone. The goal of my project is to create a central database of gasoline prices in my area, which will be published online. I would like to have an automated system place a call

G'Day All, Newbie here. How can I go about troubleshooting a fast busy when I dial my the phone number on my * server? Thanks. Ferg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041012/a669d795/attachment.htm

...eems to be in the DB but * doesn't seem to see anything after a reload even though it is acknowledging a load of x.conf (where x is extensions/voicemail) upon reload. The same files work great in flatfile form. What am I missing here? Running cvs-v1-0-11/26/04. Thanks, Jason Kawakami www.optellabs.com Salt Lake City, UT