similar to: AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFO problems

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

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

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

I have read the user reports on the Aastra 480i SIP phones, but the last report is from October, and the advice was to stay away from these units. I was looking at the cisco 79XX series or the SPA-941's, but they don't have backlit displays. I also like POE support so I don't have to run multiple wires to the phones and makes for a neater look. I dislike the looks of the SNOM

Hello, I've got my dialplan configured to do a double ring when a customer service call comes in, and a normal ring when an extension is dialed directly. When a customer service call is transferred, I want to ring to revert back to normal. In the local extension macro, I have the following ; make sure ring is set to default exten => s,n,NoOp(${ALERT_INFO}) exten =>

A call comes in I set the distinctive ring by setting variable ALERT_INFO then dial a SIP channel. The channel is answered, but then the user forwards the call to another SIP channel. ALERT_INFO is still set. How can I clear the ALERT_INFO variable after the SIP channel is answered so that when the call is forwarded the ring goes back to "normal"?

Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

Hi, I've just found: http://lists.digium.com/pipermail/asterisk-users/2003-June/014475.html which talks about ALERT_INFO and Cisco phones. How do I actually get this working and what does it do? Do I need to add anything to the configs for the phone or is it just a SetVar(ALERT_INFO=1) - which I tried and it seemed to do nothing at all.. Thanks Andy

Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I

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

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

Hello, I have uploaded a custom ringtone to our SPA-922's for the purpose of sounding like a door bell chime when the doorbell is pressed. I am using __alert_info to set this ringtone. It appears that I can only set the ringtone via alert_info for the ringtones that come from Linksys. Has anyone else seen this issue? I am doing the following: exten => 100,1,SetVar(_ALERT_INFO=doorbell)

Is asterisk able to forward it's ALERT_INFO data to another asterisk server ? My situation should look like the following: Call comes into asterisk1 in SIP. Asterisk1 sets the ALERT_INFO=Bellcore-r2, Asterisk1 dials Asterisk2 (SIP), Asterisk2 dials our SIP device which should ring with the Bellcore-r2 Any way to pass the ALERT_INFO through to the SIP device? Thanks -- Benjamin

I am impressed, I have been trying this for sometime using the SIP image and the only difference I can create is a 'single' and a 'double' ring on the phone. I use the 'single' ring for phone calls and the 'double' ring for the doorbell. I would love to be able to choose a ring tone based on the incoming msn or callerID. The idea of the phone shouting 'Its the

Hi there, see subject: Am I right that with current CVS "ALERT_INFO" doesn't work anymore? FYI: This is supposed to select the ring tone on a Cisco 7960. Cheers, Philipp

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good evening, I have something strange, when I add an ALERT_INFO variable to a ring group, the invite generated contains 2 lines with Alert-Info and my phones return a 400 Bad Request... I've checked in my config files, there is only one line with Set(__ALERT_INFO..... Any idea?? PS: I'm using Asterisk Asterisk 1.4.13-BRIstuffed-0.4.0-test4

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

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): ;