similar to: ADSI & Vista/Aastra 350

I have an Aastra 390 ADSI phone. It's not locked. I can call ADSIProg without a problem and it programs my phone. Calling Voicemail2 also programs my phone. However, in order for the VMail option to appear on the screen I have to go into the Services menu, pick Asterisk PBX and pick Select. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the

Has anyone solved the problem on the ADSI phones that when you hit one of the soft keys, the Number Pad stops working?

I've received my Aastra 390 phone. I got the unlock procedure from the vendor, and the services button now shows all four entrys as "<available>". When I give the phone "ADSIProg()" the phone displays: Asterisk PBX download refused Conflict with: <available> The asterisk.adsi hasn't been changed: DESCRIPTION "Asterisk PBX" VERSION 0x02 SECURITY

I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

Hi, It is possible to create a dummy account (SIP or IAX type) in order to be used in a "dummy" extension? I want to be able to use it as a normal extension (as an IP phone connected to it), but without the need to answer or call from that extension. I want that when I call that extension to hear the ring, and after the defined period of time to enter in the Voicemail system. I

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

To whom it might concern, The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Price is 77 euro something and available with Telec. (NL) Met vriendelijke groet, Pauline Middelink -- GPG Key fingerprint = 2D5B 87A7

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?

hi, i have to do a demo with asterisk, unfortunately i don't have yet an x100p card, so i need to use a 56k voice modem on my motherboard... could someone tell me how i can configure asterisk to use this modem to call? thanks a lot for the help!!! Angelo

I've seen this happen a few times and I think it's when the system that Asterisk is running on crashes due to a power failure (or for some other reason that causes a non-planned shutdown). While Linux comes up fine, Asterisk won't start because the drivers are loading in the wrong order. fixed by: 1) sh /usr/src/fix-asterisk-modules.sh 2) sh /etc/init.d/asterisk start Is this a

Hi List, I am trying to locate some detailed documentation and sample configs. I downloaded and compiled Asterisk, and I haven't been able to find much detailed docs on the config files. The distribution I compiled and installed doesn't have any config files, and the handbook is good but doesn't cover all of the configs. Here's my specific problem, when launching Asterisk for the

Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n" "INSTALLED FOR CONFERENCING FUNCTIONALITY.\n" I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What

In this message I try to summarize what I have learned in these last two weeks. My primary sources of informations were the * list archives and linux ISDN docs. I ain't no * master, so don't trust too hard. Relevant messages from the * list for the current discussion are: 009177.html 009268.html 0498.html 0849.html My setup is an Eicon Diva (HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2)

Hi All, I recently purchased the Asterisk Developer's Kit (TDM) to try out Asterisk. After following the directions in the Digium's FAQ topic entitled "Q. How do I configure my TDM40B and X100P?", I'm receiving the following error: WARNING[1074428608]: File chan_zap.c, Line 6748 (load_module): Ignoring rxwink ERROR[1074428608]: File chan_zap.c, Line 6692 (load_module):

Hi all, I can't seem to find any info on this anywhere on the web, except that BT caller ID doesnt use the standard bellcore system in use in the US. So, if anyone here in the UK is onlist and using the x100p successfully, please let me know. TIA, Dave

Hi All, I have an equipment loaded with 4 X100P (numbered 1-4)) and one T400P (numbered 5-8). Everything works fine except that I cannot use one of the FXS ports (number 5). If I configure zapata.conf to recognize it, the whole system voice quality suffers. I've tried already to switch PCI slots, with no results. Below is a snapshot of my /proc/interrupts, maybe this can shed some light on

Hello, I've seen various suggestions thrown around for hardware when people ask, but can we all agree on some basic hardware recommendations for a few basic setups(and post them on a website) to make it easier for new people to avoid some of the hardware/software pitfalls when they are setting up their first systems. Something like this: (THIS IS JUST A PROPOSED LAYOUT SO PLEASE BE GENTLE)

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

Program your locked ADSI phone not by unlocking it but by using the feature slot FDN download descriptors of ADSI SEC security code telco provider lock number key from database list. ch33s3 First reset phone. Set clock to Jan 1 12:00AM (options-2?). exit to main screen. press options, (mute/flash/speaker/?) until you see firmware version. press # to reset. google more info, bam ch33s3d ;