search for: antkowiak

...ion. Does this cool beast exist? And if so, where might I find some docs on it? Also, I noticed there was some talk about a gnome interface to asterisk, does that exist as well? If so, same question... Thanks in advance, and thank you to everyone who have answered my questions so far. --- Joe Antkowiak CTO, JetStream Communications, LLC

So, I can't figure out how to get the parkandannounce application to work the way I want it to... I have cisco 7960 IP phones using SIP, and this is what I have in my extensions.conf: exten => 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${EXTEN:1},1) exten => 700,2,Hangup and in my parking.conf: [general] parkext => 700 ; What ext.

Hi, I am using a channel bank and zaptel hardware. I have a credit card machine on one of the channels that appears to be dialing "too soon" for asterisk, every complete number recognized by asterisk is missing the first 1-4 numbers. This is a serious problem for me, anyone have any ideas on whats going on? The pstn picks up on the dtmf tones just fine. I was able to get it to

...titech. The echo cancellation on this device is amazing. There is no trace of the echo and the conversation is still full duplex. And, the box works perfectly with asterisk. Unfortunately, the retail price on these boxes is $3k. Just thought I'd share my experience... -- ____________ Joe Antkowiak antkojm1 (at) gmail.com

Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

I have this in extensions.conf: exten => 1,1,Dial,Zap/g9/4439568899/|24 exten => 1,2,VoiceMail(u8004) and this happens: -- Playing 'js-joe-trvm' -- Executing Dial("Zap/2-1", "Zap/g9/4439568899/|24") in new stack -- Called g9/4439568899/ -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Nobody picked up in 24000 ms -- Hungup

I noticed a few other messages posted about this problem, but I couldn't find an answer... I'm having a problem with SIP echo when calls are received into asterisk via an x100p and bridged with a sip extension (back to the pstn with iconnecthere). the person calling in to asterisk has no echo problems, but the recipient of the pstn call, everything they say, they hear back about 1 second

Ok, so I figured out my problem with my pt480s. But, now I have a few more. 1. When I dial into the voicemailmain or voicemailmain2 application, the phone and * start talking adsi, but then the phone tells me "programming download canceled, services is full.", but my services list isn't full, only "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas?

Are there any known issues with putting 2 4-port T1 cards in a single box and having all ports and all channels in use at the same time? Planning on 4 of these boxes, dual AMD cpu MB from MSI, 512m, redhat 9, agp video, on board NICs, serial ata raid. Any input would be appreciated. -Joe

Hi, I have a 1 port T1 card in an asus p4p800-vm board with a 2.0g Celeron, and 512m of 266mhz ram (256 on each channel). This board has video, Ethernet, and serial ata all on-board, I got it because of that, there wouldn't be anything else on the pci bus that would mess with the zaptel card(s). So, here's my problem. I'm running redhat 9 with kernel 2.4.20-20.9. I can load the

When I installed my first home-PBX three years ago, I was looking at "cellsockets" -- devices which will accept certain cellular phones and provide an RJ11 jack, generating the ring-voltage and recognizing DTMF, which in turn makes your cell-phone look like a CO line. Pretty cool stuff, in theory, but it just didn't seem to be worth the cost, especially since it locks you to a

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten => s,1,Hangup (I'm just using the line for outbound dialing) -----Original Message----- From: asterisk-users-admin@lists.digium.com

digium support answered, for anyone's future reference, they should be left unlocked (aka set to all zero's). Thanks. -----Original Message----- From: Joe Antkowiak <joe@jsci.net> Date: To: asterisk-dev@lists.digium.com <asterisk-dev@lists.digium.com>; asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> Subject: [Asterisk-Dev] adsi phone codes? Does anyone know off hand what codes should be used with the adsi phones in order t...

...FDN Security Code 3) Suspend Lock Flag 4) Delete lock bcus in my experience only the mfg aastra can do this >digium support answered, for anyone's future reference, they should be left unlocked (aka set to all zero's). > >Thanks. > >-----Original Message----- >From: Joe Antkowiak <joe@jsci.net> >Date: >To: asterisk-dev@lists.digium.com <asterisk-dev@lists.digium.com>; asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> >Subject: [Asterisk-Dev] adsi phone codes? > >Does anyone know off hand what codes should be used with the ads...

...ssage----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jayson Vantuyl Sent: Friday, May 30, 2003 9:12 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] aastra pt480 and adsi On Thu, May 29, 2003 at 08:58:43PM -0400, Joe Antkowiak wrote: > Ok, so I figured out my problem with my pt480s. But, now I have a few more. > > 1. When I dial into the voicemailmain or voicemailmain2 application, the > phone and * start talking adsi, but then the phone tells me "programming > download canceled, services is full.&q...

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

Hello, I am trying to make a call from our PABX to Asterisk on PRI interface. How can i configure Asterisk to enter the overlap receiving state if the complete number is not obtained in setup message. Looking forward to any help in this regard Regards Nauman Bin Ali __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection