similar to: Clear ADSI Configuration?

I was playing with a Sprint Vision phone recently and noticed when viewing the low level ReadyLink configuration screens that there are references to SIP registrars and the like. Does anyone happen to know if Sprint's implementation of ReadyLink truly is SIP based, and if so, managed to get it to interoperate with Asterisk. If so, it would prove to be an interesting paging mechanism and

Greetings, I have seen a few postings in the past regarding the interop of Asterisk and the Cisco 7920 WiFi phone. To date, I have not seen a definitive method to getting the phone working. Assuming someone has this actually working, can that person step up and answer these questions. 1) What Channel is it working with (chan_skinny or chan_sccp)? 2) If code was used that is not a part of a

I haven't seen anybody so far post a complex fix for the udev problems on FC3 with the latest kernel. On that note, I have a temporary fix to allow zaptel to load somewhat normally. I found that modifying the zaptel script to 1) load, unload, then load the driver modules and 2) insert a pause between modules seems to allow things to work. This assumes you have followed the instructions

Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code)

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

Bill I have adtran750 with Vista/Aastra 350, 390, & 480's never seen that one tim -----Original Message----- From: asterisk at billheckel.com <asterisk at billheckel.com> To: asterisk-users at lists.digium.com <asterisk-users at lists.digium.com> Date: February 21, 2003 12:52 PM Subject: [Asterisk-Users] Fun new problem with Aastra 390 ADSI phones >I seem to have

Hi all, I've been told that part of best practices is to noload all modules that you don't use. Adsi being one of them (are there actually people using adsi & asterisk?) I noloaded it in /etc/asterisk/modules.conf: noload => res_adsi.so noload => app_adsiprog.so When I start * (this is cvs head of today by the way) I get: [app_voicemail.so]Aug 31 17:25:49 WARNING[4961]:

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

Folks, I am in the process of fighting my way through my first foray into ADSI phones. I have a very nice Aastra 480 from Sayson, purchased through Cylogistics. Very nice phone. However, my experience in configuring ADSI has been pure pain so far, and it's not all working yet... 1) The locking mechanism for these ADSI phones is... baroque? Arcane? Perhaps there is a reason why you would

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,

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.

I just need to buy 5-6 ADSI Phones. I wondering which of these models to choose - aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and without ADSI - aastra 350 - I'm sure that it have ADSI. If there is some other good working model, it will be great, if someone points me. Thanks

I have ADSI working on my Aastra (Vista/Nortel) 350 phone and everything is working fine. However, I want the asterisk.adsi to load into the 'self-load' slot but can't figure out what the correct FDN for doing this is. Does anyone know the right FDN for the SL slot on these phones? Also, does anyone have any cool/interesting ADSI scripts they wouldn't mind sharing? I'm

This is a really cool phone, except one problem, searched the archives and this was brought up before. Just wondering if anyone figured out how to solve it. I'm having the same problem as these previous posts... --------------- posted 06/09/03 Whenever I try using the voicemail through my ADSI display, it disables my # buttons. If I hit listen through the ADSI display, I can not delete

My set up is quite simple: Redhat 9, Asterisk CV-HEAD from 07/11/04-13:04:22; TDM400P card, FXO card (X101P now unused) and a Fritz! AVM pci ISDN card. It works. But...just got an adsi PT390 phone and with some help from the nice people at Sayson got some basic programming into it. The programming came from dialling over a traditional phone line however and not from Asterisk. Whenever I

Hey Everybody, You may remember be as the one that was generally running in circles trying to get ADSI phones to work? Well, five phones later, I've got a phone and a code that works to program it. I have been discouraged from just posting it to an e-mail forum, but I have been encouraged to get it embedded in the program. Is this possible? How should it be done? Next, after getting it

I was wondering if anyone has had any experience with PowerTouch 350 & ADSI. I am able to configure the phone with ADSI.. Which is pretty cool in it self. But I have a question regarding the Service feature. I have to choose the service button and choose the Asterisk PBX ADSI. But let's say I make a call it does not default back to this. I have to go back into Services and choose it

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): ;

Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia,

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?