similar to: Re: Asterisk and MWI

Thank you Melissa. I love the phone but the dial keypad is a little bouncy. I was hoping for a more solid feel like on the analog PT390's or my quality standard, the Nortel 9417CW. Other than the MWI problem, I'd like more documentation on the configuration paramters. I have found little online configuration documentation other than very basic stuff on the Sayson website. I'd

But where do can you get this later firmware from? I'm still on 1.0.0.78 on my 480i. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter Passchier Sent: 05 August 2005 00:04 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems

Hi rsync list! I have a problem with the exclude-from and include-from options. I want sync all jpeg files to an external server - exclude HK and texts folders and jpg inside this folders. My folder structure are: /products/ANYNAME/pictures/EU /products/ANYNAME/pictures/HK /products/ANYNAME/texts ... my include-from file: */ *.JPG *.jpg my exclude-from file: */HK/ */texts * my rsync

Hai, Whats the os used? The first things i would check. Did you give both servers the nfs/spn. ? The current search order for keytabs to be used for "machine credentials" : <HOSTNAME>$@<REALM> root/<hostname>@<REALM> nfs/<hostname>@<REALM> host/<hostname>@<REALM> root/<anyname>@<REALM> nfs/<anyname>@<REALM>

Hi all. I am a total newbie at Samba. I am running Ubuntu 6.06 (Dapper) and have Samba 3.022 installed. I am trying to set it up as a Domain controller (to be a domain server that Windows machines can log into). So far, I can get it to do this, but only with the ROOT account. Any other account and it tells me that the user/password is incorrect or invalid. Below is my smb.conf file. In the

Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

I have a strange problem with my Sayson 480i IP phone. If I press the Transfer button and then dial extension 200 to try to transfer the call, the Sayson apparently is treating the 200 as the last part of an IP address, and the call fails. As soon as I enter 200 and press Dial, I see an IP address of the form x.y.z.200 show up on the display. How do I get the phone to treat the dial string as an

We have a customer with a handful of Sayson/Aastra 480i phones behind a Juniper Networks Netscreen firewall registering with our hosted PBX service. The Netscreen monitors the REGISTER messages and only keeps the reverse mapping open for the duration of the registration period. It appears that every so often the Sayson does not send out another REGISTER message after the registration has expired

I'm currently testing these phones and so far I'm very pleased with them. I'm using the SIP image (Version: 1.0.0.23 Release: 0032-01-A4). There are a few issues with the speaker not working properly when you dial out or pick a call but this is a known issue and will be hashed out in the next release. I've been working with the developers at Sayson and they have been very

Asterisk 1.0.3 Sayson 480i running .78 release (problem may not be Sayson specific, it's just that's what's deployed) Problem: Asterisk rejects registrations every so often even though nothing has changed either with Sayson or Asterisk configuration (and previous registrations have succeeded) SIP trace of successful registration: =============================

For those that are interested, here is my report back to Sayson on the 480i ---------- Forwarded message ---------- Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT) From: xxx@bgcfreedom.com To: xxx@sayson.com Subject: 480i User Feedback With Asterisk Seshu, I am using a 480i, and I am impressed with the phone on a whole, but obviously the firmware is lacking. Details follow. Hold button works, but

Hello, I have been wanting to try a Sayson 480 ADSI phone with our * box, but I haven't had much luck getting the phone to use *'s built in ADSI script. Does anyone know if there are any how-to's out there for this? Or, could anyone enlighten me? Thanks, Craig --------------------------------- Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. --------------

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

Let me first admit: I am experiencing this bug on a precompiled binary of R for windows, and I know these are not supported. I am not asking for support, but I thought the R developers would want to know about this. # Define: foo.page <- function(x) x # Then, foo.page("a") # actually invokes the pager (!) on object named "a" (if one is defined). # same thing happens

Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? All I get, no matter what I've tried is "Unable to upgrade firmware". tftpd is working because the dialplan freshens, and I have aastra.cfg whatevermacaddressfile.cfg and firmware.st in /tftpboot Am I missing something stupid? Is there another way to upgrade it? Chris Coulthurst chris@shuksan.com

Hi all, I'm experiencing a problem with SIP channels going ZOmBIE after the following sequence of events: - IAX2 client calls SIP client - SIP client consultive transfers (using sip REFER) the call to a MeetMe extension, and hangs up. At this point, the IAX2 client will indeed be in the meetme room, but a 'show channels' at the * CLI reveals that the SIP channels that were involved

I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com <http://www.successfulhosting.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041128/8e282c51/attachment.htm

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

Still looking for cheaper (under $250,-) alternative to cisco 7940 with features needed for corporate use, mainly: - shared phone book (e.g. via LDAP or XML browser in phone) - in-line power - missed/dialed/received numbers - integrated switch (voice VLAN support) I found only aastara/sayson phone (and Intracom/Netphone in the past), that has xml services anounced, but still not available, so

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): ;