similar to: ASDI Programming through an ATA/SIP device?

Greetings, I have just purchased several Astra 390 phones ready for asterisk. I have placed a line with adsi=yes in the Zapata.conf file just before channel => 13 I have also added an extension exten => 6199,1,ADSIProg(asterisk.adsi) exten => 6199,2,Hangup in the extensions.conf file. When I try to program the phone I get the following: Asterisk CVS-03/28/04-12:02:10,

Hello all, New to the list, Asterisk and this wonderful world of convergence. From reading the docs, list and websites I hopefully correctly understand that I can use a plain cheap isdn4linux supported ISDN (passive) TA to hook up the PC to the ISDN line from the telco (FXO card?). Next in the PC I also need a FXS card like a Digium TDM10B or PhoneJack so I can hook up an analog phone to it. In

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

Can anyone tell me what I need to do to tell the voicemail and voicemail2 applications to download to a different slot? The only problem I have remaining is this one... I dial the voicemailmain extension from a PT480, and the display tells me "services is full, download refused", but asterisk PBX only occupies the first slot (I changed this)... Any help would be greatly appreciated.

Still I cannot get ADSIprog to work, I have tried everything. -- Executing ADSIProg("Zap/1-1", "") in new stack -- ADSI Available on CPE. Attempting Upload. == Spawn extension (co1, s, 1) exited non-zero on 'Zap/1-1' * Hungup 'Zap/1-1' It hangs up right after it says it is attempting to upload. -------------- next part

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): ;

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

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?

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

What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the

**** DRAFT **** DRAFT **** DRAFT **** DRAFT **** I've been using the DTA310 from Packet8.net for a couple of weeks. The DTA310 is about $130 without the Packet8.net VoIP service. It only supports SIP. On the back of the DTA310 is a power connector (power supply is provided with the product), a 10/100 Ethernet port, an FXS port, and a reset button. The front of the device has LEDs for

I sent this yesterday from my wrong email account. yeah, Thank you. I screwed up by only looking at adsi.conf, and not asterisk.adsi. Once I saw that, things started falling into place. Are you able to pass arguments to the phone so that for every call there is unique data on the screen. I see that works for caller id with $Call1p and $Call1s, but what If I obtain an OrderID from my

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

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems

Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up Asterisk (with -cvvv) I get an error regarding func_curl.so (lines omitted) ... == Registered custom function STRFTIME == Registered custom function STRPTIME == Registered custom function EVAL == Registered custom function KEYPADHASH == Registered custom function SPRINTF func_strings.so => (String handling

I found some old messages regarding a possible pkt8 DTA "bypass". Anyone is using Packet8 with Asterisk? ========== http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat Got Softphone Working with Packet8 Friendly name: {Anything you'd like.} SIP domain: packet8.net SIP proxy: packet8.net Leave everything else at the default. When you login, it will ask for a username

Looks like www.packet8.com <http://www.packet8.com/> has been hacked :-( The phone service is offline as well. Interesting to note that the whois is showing an update today but doesn't look like details have changed. Domain Name: PACKET8.NET Registrar: REGISTER.COM, INC. Whois Server: whois.register.com Referral URL: http://www.register.com Name Server:

Dear List I would like to ask you something concenting a better print of the R output: I have a bit data frame which has the following structure: CFISCALE RAGSOCB ANNO VAR1 VAR2......... 9853312 astra 2005 6 45 9853312 astra 2006 78 45

I'm trying to get asterisk working w/ packet8 (incoming and outgoing) and a dta310 so I can have more control over voicemail. I've looked at the data stream coming from the dta310 and from packet8, but I haven't managed to get the phone to actually place invites through asterisk. On the asterisk end with chan_oss.so, I can make it dial and I hear ringing and the first second of

Hi John, I've been using Packet8 via a physical ATA and XP100 card for some time. As far as I know it is not possible to connect to the Packet8 service without the ATA. If this is not the case I would be very interested to hear this. In addition since moving to the USA I now only have a single packet8 line into my asterisk box (I used to have this and a 2nd regular pstn line) I used to be