similar to: OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)

Cisco phones act a as a switch. If you do not use the CDP protocol to "tell" the phone it needs to be in a special VLAN (802.1q) then it will just use the access port settings on the switch, and, also allow the PC connected to the 2nd Ethernet port to have access to the network. However, if you have an all cisco powered network, with all cisco phones, I could advise you to use the CDP

Switch is only tagging the vlan packets. Once the PC loads the vlan aware driver ("client") it will be able to read tagged packet for the vlan which PC has been configured to use. Nothing to be done on the switch. W -----Original Message----- From: Joao Pereira [mailto:joao.pereira@fccn.pt] Sent: Thursday, March 02, 2006 1:15 PM To: Wojciech Tryc; asterisk-users@lists.digium.com

On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote: > Does anyone have a way to do wake calls? > > > > Jordan Novak > > Communications Technician > > Logistics Health Inc. You could use cron and /var/spool/asterisk/outgoing scripts to dial numbers, etc... > Can you elaborate, I am fairly new to Linux and a phone guy to boot. I am looking for a way for the

I have recently upgraded my firmware from v6.3 to v7.4. Now when the phone is booted or rebooted, the initial screen "Initializing Vlan" takes forever to initialize before it initializes IP. Any ideas/Thoughts? (Trying not to Revert back to v6.3). -C -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, I have to configure Xen server and DomU''s which should be able to talk to different vlans - depending on services put on DomU''s. I would like to enslave 2 NICs via bond interface. Both NICs are on trunk ports which are configured in the same way, the same native vlan. I don''t know, if I have to add vlans tagging to eth* interfaces then enslave tagged eth*

Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close

I am using 8021q vlans on my servers, and I cannot figure out how to make PXE/DHCP work. Has anyone figured out how to do this ?

Hello all, I have a CentOS box that has a NIC (eth0) on which I defined 4 VLAN's (counting the NIC itself): eth0, eth0.1, eth0.2 and eht0.3. Initially the Cisco switch was not partitioned into VLAN's which means that the only VLAN running on it was the default one (VLAN 1). I have then played with VLAN's a bit on the switch and at this point have two: VLAN 1 (which is default and can

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten => _555,1,ChanSpy(Agent) exten => _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which

Do you need the whole configuration? On the switch end, we have the relevant VLAN (VLAN 48) with the assigned IP address of 192.168.48.101 and the range of ports (Gi1/0/1 - Gi1/0/8) assigned to that VLAN. Seems - and acts - like a legitimate setup and works fine, except for this particular instance. Thanks. Boris. On Fri, Jan 23, 2015 at 8:54 PM, Dennis Jacobfeuerborn < dennisml at

I figured I should post this because before I went to do it I searched the web for caveats etc and didn''t really find much. So anyway, I set up bonding + vlan for a project. Computer is dell powerdge 350 with 2 intel 82559 10/100 nic running stock 2.4.20. I tried both the eepro100 and the e100 drivers for the nics, both work 802.1q and bonding are modules. Switch is cisco 2924xl-en

Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao

Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira

Hello I'm using eyeBeam, and Asterisk keeps sending my clients a second call, when they are still in one call (because eyeBeam has lots of channels). I was using X-Lite (with 3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao

Hello I'm trying to use an AGI that returns the queues status (numbers of available agents, etc ), but I'm having some problems with it (it's still very buggy). Is there any AGI repository with source code samples? Had anyone used an AGI to check queues and agents status? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip:

Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:password at sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Thank you everyone. OK, the mystery deepens, I guess. The machine does need to support several VLAN's, it is currently on a trunkport (8021q encapsulated), it made it into the ARP table - which I specifically tested for by physically unplugging the table, clearing the ARP table and plugging it back in. The ARP table currently looks like this: hq#show arp Protocol Address Age (min)