search for: jkkawakami

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

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

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

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have

----- Original Message ----- > Subject: Re: [Asterisk-Users] Analog Phones with Status Light Indicators > From: Adam Goryachev <mailinglists@websitemanagers.com.au> > To: asterisk-users@lists.digium.com > Organization: Website Managers > Date: Thu, 12 Aug 2004 14:53:02 +1000 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-08-11 at 20:42, Steven

...gup > exten => 101,102,Voicemail(b101) > exten => 101,103,Hangup > > [home] > include => fwd-out > include => local > include => long-distance ------------------------------ Message: 3 Date: Wed, 22 Dec 2004 19:43:35 -0700 From: "Jason Kawakami" <jkkawakami@optellabs.com> Subject: [Asterisk-Users] RE: Zaptel/Zapata config from T410p to Brooktrout T1 To: <asterisk-users@lists.digium.com> Message-ID: <200412230239.iBN2dQQZ054184@dkh.dsl.xmission.com> Content-Type: text/plain; charset="us-ascii" -----Original Message----- M...

...etVar(ALERT_INFO=Bellcore-dr1) > > And that works fine. > > What was the error message you were getting? > > -- > -- > Sam Tilders > sam@jovianprojects.com.au > (Move to Jupiter) > > --__--__-- > > Message: 2 > From: "Jason Kawakami" <jkkawakami@optellabs.com> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration > Date: Mon, 19 Jul 2004 17:28:09 -0600 > Reply-To: asterisk-users@lists.digium.com > > Date: Mon, 19 Jul 2004 14:54:44 -0500 > From: &q...

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have "orphans" around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I

Trying to use IAX2 extension as call center agent but getting this on the CLI May 24 20:34:20 WARNING[1209214400]: channel.c:1783 ast_request: No channel type registered for 'IAX2[2001@2001]' using AddQueueMember as the login mechanism and that seems to work but * will not deliver to the IAX2 extension. any ideas? Jason Kawakami -------------- next part -------------- An HTML

> Message: 13 > Date: Fri, 9 Jul 2004 11:42:01 +1200 (NZST) > From: =?iso-8859-1?q?Eugen=20Cristea?= <tecristea@yahoo.co.nz> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] asterisk to asterisk config > Reply-To: asterisk-users@lists.digium.com > > Hi, > > I would like to set two separate asterisks to talk to > each other. > Any

-----Original Message----- I just want to clarify a few things. I have about 100 Toshiba digital phones, 4 ports on the voicemail, and 24 phone lines. Not all of the lines are POTS lines. I think 8 of the lines are Direct Inbound Dial (DID). Due to a decrease in call volume, I am most likely going to cancel 6 - 8 POTS lines in the near future. Also, due to the cost involved with implementing

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am 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

First time doing a t-1 config with *. running FC2 and CVS as of last W/E. i run modprobe wct1xxp and I get "Unable to open master device '/dev/zap/ctl' i look at /dev and there isn't even a zap directory. i was looking around and found something in /udev that looks promising but not sure what the deal is. ztcfg returns 24 channels configured but the same notice at the bottom

is there something that needs to be set up to make the 'called' and 'callers' buttons work on this phone? all i get is the backlight to switch on and off. Jason Kawakami

started messing around with the Monitor cmd for call recording but noticed that the cmd was 'injecting' a small noise every 8 sec or so. looked through the wiki for a flag setting but the 'm' for using soxmix post call is the only one noted. trying to build a call logger that just sits next to another switch and does the recording but if it gives an indication that it is

----- Original Message ----- > From: Greg Hill <gregh-asterisk@hillnet.us> > Subject: Re: [Asterisk-Users] New to * > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <Pine.LNX.4.44.0409031231070.1975-100000@hillnet.us> > Content-Type: TEXT/PLAIN; charset=US-ASCII > > On Fri, 3 Sep 2004, Bill

----- Original Message ----- > Message: 3 > Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT) > From: David Troy <dave@popvox.com> > Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over > National or DMS100 PRI to a Norstar MICS? > snip> > > I have a PRI link up and running between Asterisk and a Nortel Norstar MICS > > v4.1 . I'm having a