similar to: Codes negotiation problemsbetweenAsterisk1.4beta2 and Aastra 480i

Hi! I am in the process of upgrading our 1.2 servers to 1.6. We have a lot of realtime extensions with app=Macro and appdata=stdexten|080512|SIP/080512 But this does not work in 1.6. It is expecting , and not | as the argument seperator. If I change the | to , then it does not work in 1.2. Is there any backwards compatible switch you can enable in 1.6, so it accepts | as a argument seperator

Hi! We have some problems with calls that get disconnected in the middle of a call. We are using Asterisk 1.2.1 with a TE410P (2.gen firmware). When the call is disconnected Asterisk writes this to the log: Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy, avgtone: 300, avgsilence 2090 Jan 9 14:56:17 DEBUG[4404] dsp.c: Requesting Hangup because the busy tone was detected on

I am installing Dovecot-2.1.13 on Slackware 13.37.0 for the first time to replace ipop3d and I have overcome all of the roadblocks as they have developed except this last one and I finally have to say ?uncle?. The error messages that are showing up in the dovecot.log are as follows. Jan 24 12:27:27 tux2 dovecot: auth: Error: passwd-file /etc/shadow: open(/etc/shadow) failed: Permission denied

The way to do this is with ACL's. Access Control Lists IPtables can perform this function, or an internet gateway router can also be used. The ISR 4000 Series Cisco router family is where I would start, especially if you're in the need for a blade server in the same chassis. -----Original Message----- From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Nicolas Kovacs Sent:

Hi, In our local school we have two servers and roughly 80 clients. The network is 192.168.10.0/255.255.255.0, and DHCP+DNS is managed by Dnsmasq. School PCs (teachers and management) are registered via MAC address and get an IP address in a specific range: 192.168.10.2 - 192.168.10.50 - management + teachers 192.168.10.201 - 192.168.10.220 - computer room 192.168.10.246 - 192.168.10.247 -

Hi! We have a SIP server with a TE410P card with asterisk version Asterisk CVS-D2005.02.12.14.37.11-04/13/05-16:14:03. Sometime the calls get disconnected with now reason and the users get a busy signal. The log file show this for one of the calls that got disconnected: Aug 31 22:51:53 VERBOSE[3911]: -- Accepting call from '46362302' to '36917474' on channel 0/5, span 1 Aug

Hi all I have two samba PDC installed according to these specifications: domain FOOBAR with pdc server name: BAR (ip 192.168.1.1) opensuse 11.1 samba-3.5.6-15.1 openldap2-2.4.12-5.6.1 smbldap-tools-0.9.5-25.1 A winxp called USUARIO1 joined to the FOOBAR domain (ip 192.168.1.100) domain LAPAZ with pdc server name: SERVERLPZ (ip 192.168.10.4) openSUSE 12.2 samba-3.6.7-48.12.1.i586

[asterisk-users] Asterisk <-> Nortel Phone Switch Date: Thu, 29 Nov 2007 07:52:17 +0000 (GMT) X-Mailer: sendEmail-1.52 MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="----MIME delimiter for sendEmail-20854.4017086787" This is a multi-part message in MIME format. To properly display this message you need a MIME-Version 1.0 compliant Email program. ------MIME delimiter

Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no

Hi, We're having problems dialing out to Asterisk from our Grandstream GXP-200 phones. About 2 of 3 times, when we dial, nothing happens. Looking at the console in max debug mode, there are no messages except the following: Feb 24 10:29:20 WARNING[2475]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 9913b47bcd7 aeb52@192.168.10.100 for seqno 4524 (Critical Response)

Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no

Hi All: I have a Cisco3620 with a proper T1/PRI card installed with asterisk running on the same LAN. Since I have lit up the line, I can dial out and make calls to regular lands lines. However when a call comes back in it rings the destination phone once and disconnects. Here is an error from my router 15:40:45: ISDN Se1/0:23 SERROR: L3_GetUser_NLCB: EVENT 0X45 No NLCB 2 15:40:45: ISDN

Package: xcp-storage-managers Version: 0.1.1-2 Severity: important I still am not able to create iSCSI storage repositories using this command xe sr-create host-uuid=<my-host-uuid> content-type=user name-label="LVM over iSCSI SR"> shared=true device-config:target=192.168.10.100 device-config:targetIQN=<here-target-iqn> device-config:SCSIid=<my-scsi-id>

I have been playing with a 480i with the new firmware 1.2.0.162 I hope to get some form of paging intercom function to work. In the wiki someone post that ALERT_INFO type of paging might be in this version of firmware but I have been unable to find anything on this yet. I have tried sending the ALERT_INFO to the phone a number of ways with no results. I then hooked up my bt100 and tried to dial

Having scoured the web, I still am no better off .. I have 2 Aastra480i's , and 120+ cisco 7940's :) . I am trying to decide which model to use going forward when we purchase more kit. They both seem much on a par regarding features. Q1: Is there anyway of making the cisco auto-answer _without_ having to manually edit the configuration on each phone ? I've been able to get the

Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Can't seem to find any info on the Aastra site. Comments?

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

After upgrading to 1.8.3.2 today, I notice that my Aastra 480i SIP phones no longer initiate hold music when a call is placed on hold. I seem to be having the same issue as the person here: http://forums.digium.com/viewtopic.php?f=1&t=77553 Has anyone else run into this issue? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC

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