similar to: RE: Asterisk-Users Digest, Vol 20, Issue 13

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

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

I need your help I have a sangoma A104D on my dell server; I got card status ok with no alarm If I dialed the extension 6210006, it shows the output as stated below, but there is no ringing from the pstn number nor the iax softphone am using on my pc. I will be glad if someone can give me a working config? What I want to achieve is to send all my call to the pstn on A104D? The pstn am

Hello to all I would like to know If some of you have already configured an Cisco IP Phone (7940 or 7960) to work in a different VLAN than the PC that is connected through the phone switch? I know that this can be done with the Skinny firmware, but I dont if it works with the SIP firmware. The Cisco technical staff told me that these phones dont support 802.1x but can work as pass-through.

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

Hi I tried Xten, its very good, because it can stay in the taskbar (next to the clock) and start when windows starts, and is allways ready to receive calls. Maybe it s the best way to introduce VoIP to my company workers.... But theres a feature that s missing (or I couldnt find), there s no way to connect this softphone with the adress book. I think this feature is very important, because

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: Monday, September 17, 2007 12:45 PM To: joao.pereira at fccn.pt; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Center SoftPhone with Auto Answer Joao Pereira wrote: > But still, the

Hi all I have a pocketPC Toshiba e750 and I want to make SIP calls from it, but I didnt found any free softphones for my Toshiba. X lite's versions for pocketPC isnt free :( Did someone used before a free softphone for pocketPC? witch one? Thanks Joao Pereira www.fccn.pt

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we

Hello Our NT workstations always ask for the password on each share of the samba server, although the password is the same as the log on password. Is there any way to avoid this ? The first try always fails on the server, but I don't what user/passwd combination NT uses on that first time... Thanks, Joao Pagaime -- FCCN - Fundacao para a Computacao Cientifica Nacional - Tel: 351-1-8440100

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 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

Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira

Helo all I'm having performance problems with samba v2.0.3/freeBSD3.2 when there's more than one client using the same share. Every thing works, but very slowly - about 30 seconds to open an Access table. When there is only 1 client, it takes about a second. The server is almost 100% idle. I tried samba v2.0.5a on a Linux and the problem also exists. On a MSWindows NT workstation

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 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 I want to use Asterisk to implement a SIP Domain allowing my clients to do URI dialing and receive calls from the Internet through URIs and ENUM. My question is, should I put my Asterisk outside the firewall (in the DMZ) to allow connections to the Internet? Or should I have it inside my local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks

I think one of the differences is: We do pay attention to Asterisk and this mailing list ;-) CS -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Joao Pereira Gesendet: Dienstag, 31. Oktober 2006 13:47 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Snom or Cisco Phones? Hello I need to

Hi Boris, what I'd like to know is the actual VLAN configuration of the switch port (link-type and tagged and untagged VLANs). When I look at the switchport coniguration here I get (among other things): ... Port link-type: trunk Tagged VLAN ID : 8, 1624 Untagged VLAN ID : 10 ... Here is my suspicion: Your ports have an access link-type with an untagged VLAN ID of 48. That would

Hi all, Is it possible to have native VLAN (802.1q) and a VLAN x at the same time in DOMU''s? This is what I have configured on my DOM0 (xen2): Eth0 (connected by a 801.1q trunk to Cisco 2950 switch) Created vlan100 : vconfig eth0 100 Created bridge : brctl addbr br100 Attach switchport: brctl addif br100 eth0.100 Attach a DOMU : brctl addif br100 vif2.0 I now have bridge br100