similar to: Polycom getting DCHP address from wrong VLAN

Changes since v9: * series re-ordering so make functionality more distinct. Basic vlan filtering is patches 1-4. Support for PVID/untagged vlans is patches 5 and 6. VLAN support for FDB/MDB is patches 7-11. Patch 12 is still additional egress policy. * Slight simplification to code that extracts the VID from skb. Since we now depend on the vlan module, at the time of input skb_tci is

Would anyone happen to have some examples of polycom configs, specifically the 650 with sidecar for blf. I have the asterisk side all configured since I've set up blf with other types of phones, but I'm missing the polycom side. I've put together a <mac>-directory.xml, and the sidecar now lists numbers as speed dials but does not subscribe to blf.

Anyone know of a company that makes a pass through ATA? By pass through I mean have an Ethernet switch built into the ATA, like most desktop phones have. All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN ports. I fooled around with DMZ etc...but it just doesn't work as well. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, After reading XenServer documents and debugging, I found the previous configuration is correct, here is the one I've configured to work: In XCP, as in XenServer, system's network configuration doesn't really matter, it only make sense before XCP starts and handles the network settings. So there is no need to configure VLAN on host machine, and you only need a minimal one active

This is my system info: Debian Release: 10.5 APT prefers stable-updates APT policy: (500, 'stable-updates'), (500, 'stable') Architecture: amd64 (x86_64) Kernel: Linux 5.4.60-1-pve (SMP w/16 CPU cores) Kernel taint flags: TAINT_PROPRIETARY_MODULE, TAINT_OOT_MODULE Locale: LANG=en_US.UTF-8, LC_CTYPE=en_US.UTF-8 (charmap=UTF-8), LANGUAGE=en_US:en (charmap=UTF-8) Shell: /bin/sh

This series of patches provides an ability to add VLANs to the bridge ports. This is similar to what can be found in most switches. The bridge port may have any number of VLANs added to it including vlan 0 priority tagged traffic. When vlans are added to the port, only traffic tagged with particular vlan will forwarded over this port. Additionally, vlan ids are added to FDB entries and become

Does anyone know if you can have multiple TE110P cards in one chassis? -Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061011/adbceeb5/attachment.htm

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? I can push the boot server via option 66 but that is about it. I have set it for 'fixed' and tried many different option numbers with a couple differnet DHCP servers. SIP firmware 1.3.4 or 1.4.1 doesn't make a difference. Aloha, Matt

I'm looking for some information regarding the interaction of KVM, VLANs, firewalld, and the kernel's forwarding configuration. I would appreciate input especially from anyone already running a similar configuration in production. In short, I'm trying to figure out if a current configuration is inadvertently opening up traffic across network segments. On earlier versions of CentOS

We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to figure out which one of our agents is answering the calls. According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the only time the queue_log puts the channel (agent) is during logoff & logon. There is the connect & completeagent message, but it doesn't show which channel (agent) answered the phone. I can't even figure it our cross referencing the

Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I don't quite see how that works. Do you point a non-VLAN'd segment at it (akin to when you uplink a VLAN_enabled switch), and have the phone implement the VLAN? Or...? *puzzled* Thanks much, -Ken

Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog '

Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at "Auto Divert".....I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add <bw>1</bw> manually to the speed dial file it doesn't change anything. The buttons work well for

Anyone know of a SIP phone that supports group listen? Group listen allow you use the handset but what the far end says comes out the speaker...it is F802 on a Norstar. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/a219002e/attachment.htm

We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want

I would like to be able to place a call on hold via the manager interface and be able to retrieve it. The user can click a button in the Order entry form to put the caller on hold when they are looking up information. It saves them from having their hands leave the keyboard and press hold on the phone. I don't see 'hold' & 'retrieve' commands for the manager interface.

I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07

I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user