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  • How to configure Transparent IP Address Sharing (TAS) on a Mediatrix 4102 with DGW 2.0 firmware?

    - by Pascal Bourque
    I am making the switch to VoIP. I chose voip.ms as my service provider and Mediatrix 4102 as my ATA. One reason why I chose the Mediatrix over other popular consumer ATAs is that it's supposed to be easy to place it in front of the router, so it can give priority to its own upstream traffic over the home network's upstream traffic. This is supposed to work transparently, with the ATA and router sharing the same public IP address (the one obtained from the modem). They call this feaure Transparent IP Address Sharing, or TAS. Their promotional brochure describes it like this: The Mediatrix 4102 also uses its innovative TAS (Transparent IP Address Sharing) technology and an embedded PPPoE client to allow the PC (or router) connected to the second Ethernet port to have the same public IP address, eliminating the need for private IP addresses or address translations. I am interested by this feature because my router, an Apple Time Capsule, doesn't support QoS and cannot give priority to the voice packets if the ATA is behind the router. However, after hours of searching the web, reading the documentation, and good ol' trial and error, I haven't been able to configure the Mediatrix to run in this mode. Then I found a version of the manual that looks like it was for a previous version of the firmware (SIP), where there is an entire section dedicated to configuring TAS (starting at page 209). But my Mediatrix comes with the DGW 2.0 firmware, whose documentation does not mention TAS at all. So I tried to follow the TAS setup instructions from the SIP documentation and apply them to my DGW firmware, using the Variable Mapping Between SIP v5.0 and DGW v2.0 document as a reference, but no success. Some required SIP variables don't have an equivalent in DGW. So it looks like the DGW firmware does not support TAS at all, or if it does they are not doing anything to help us set it up. So right now, the Mediatrix is behind the router and VoIP works perfectly except when my upstream bandwidth is saturated. My questions are: Is downgrading to SIP firmware the only way to have my Mediatrix 4102 run in TAS mode? If not, anybody knows how to setup TAS on the DGW firmware? Is TAS mode the only way to give priority to the voice packets if I want to keep my current router (Apple Time Capsule)? Thanks!

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  • Corporate IM with video that actually works, suggestions?

    - by Erik P. Skaalerud
    Hi. Does anyone here have a suggestion for a cross-platform IM solution wich will work with voip/video on both Windows (XP and 7) and Mac OS X from 10.4 and upwards? Right now were in a kind of mixed enviroment, with some Mac users using iChat server since they need video support (conference across several offices over VPN), but it wont't work on windows clients. The rest of us are happily using Openfire+Spark, but there's no VoIP or video avaible from what i've found, unless you want to add in several 3rd party software (like red5 and asterisk). Requirements: As said before; must work on both Windows and Mac Internal server (no Skype etc) File transfer between platforms SSO (Single Sign-On) via Active Directory authentication Some sort of screen sharing would be a plus, like switching over to a screen capture (powerpoint, software training etc) We can afford to buy software if that's needed to get this working without any hiccups across platforms. Pre-thanks to anyone who gives suggestions.

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  • How to configure Transparent IP Address Sharing (TAS) on a Mediatrix 4102 with DGW 2.0 firmware?

    - by Pascal Bourque
    I am making the switch to VoIP. I chose voip.ms as my service provider and Mediatrix 4102 as my ATA. One reason why I chose the Mediatrix over other popular consumer ATAs is that it's supposed to be easy to place it in front of the router, so it can give priority to its own upstream traffic over the home network's upstream traffic. This is supposed to work transparently, with the ATA and router sharing the same public IP address (the one obtained from the modem). They call this feaure Transparent IP Address Sharing, or TAS. Their promotional brochure describes it like this: The Mediatrix 4102 also uses its innovative TAS (Transparent IP Address Sharing) technology and an embedded PPPoE client to allow the PC (or router) connected to the second Ethernet port to have the same public IP address, eliminating the need for private IP addresses or address translations. I am interested by this feature because my router, an Apple Time Capsule, doesn't support QoS and cannot give priority to the voice packets if the ATA is behind the router. However, after hours of searching the web, reading the documentation, and good ol' trial and error, I haven't been able to configure the Mediatrix to run in this mode. Then I found a version of the manual that looks like it was for a previous version of the firmware (SIP), where there is an entire section dedicated to configuring TAS (starting at page 209). But my Mediatrix comes with the DGW 2.0 firmware, whose documentation does not mention TAS at all. So I tried to follow the TAS setup instructions from the SIP documentation and apply them to my DGW firmware, using the Variable Mapping Between SIP v5.0 and DGW v2.0 document as a reference, but no success. Some required SIP variables don't have an equivalent in DGW. So it looks like the DGW firmware does not support TAS at all, or if it does they are not doing anything to help us set it up. So right now, the Mediatrix is behind the router and VoIP works perfectly except when my upstream bandwidth is saturated. My questions are: Is downgrading to SIP firmware the only way to have my Mediatrix 4102 run in TAS mode? If not, anybody knows how to setup TAS on the DGW firmware? Is TAS mode the only way to give priority to the voice packets if I want to keep my current router (Apple Time Capsule)? Thanks!

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  • Converge Voice and Data networks using Sonicwall

    - by skinneejoe
    We are looking to converge VOIP and Data traffic onto a single wire so that our client's VOIP phones pass data through to the users computer. We are specing out a new Sonicwall NSA appliance to handle routing functions and layer 2 switches to manage VLANS. Not a huge network, medium sized. What should I know about converging the networks onto a single wire? Obviously I'll want to prioritize voice traffic, is this handled solely in the Sonicwall with QoS configurations or do the layer 2 switches need to be configured differently? Any other pitfalls I should be aware of, or any good resources for learning more?

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  • IPv6 in a Small Business

    - by Martyn
    I’ve been tasked to identify a new server and IT equipment for the company I work for (employees <20). I’m looking at getting a SBS 2011 Standard box. My Question is: Apart from ensuring that our router is IPv6 compliant, is there anything else I need to do/investigate to ensure that we won’t have IPv6 related problems in the future? When investigating this I can only find references to DNS Mangers etc. The SBS 2011 box will be used for exchange and remote log in access – it won’t be hosting any websites. Also will VOIP phone systems be affected by IPv6? (We are running a rather old VOIP system, would that need to be updated?

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  • vSphere Promiscuous mode only receiving packets one way from network switch

    - by steve.lippert
    We have two network switches, a POE switch (SwitchA) to power our phones / users computers and a non-POE switch (SwitchB for the rest network.) Each switch is setup to do port mirroring to support our VoIP recording system. SwitchA does port mirroring on specific ports if we need to record a user. SwitchB mirrors one port to monitor our work at home users (Internet comes in from managed router, to switch, back out to our firewall.) These two port mirroring setups feed into one vmware vSphere 4.1 server, it has four total physical cards. The other two NICs feed into an unmanaged switch for connecting to the rest of the network. Once into the vSphere server all network ports go into a vSwitch, and then one of the servers (Windows 2008 R2) sniffs them out and does its thing. Everything is working fine and dandy from SwitchB. But on SwitchA we only receive one side of the VoIP packets (going out to the phone, nothing coming in from the phone). Troubleshooting steps I have taken so far: I hooked up my laptop to the monitor port on SwitchB and I see both sides of the packets. I swapped which network interface is plugged into the monitor port on SwitchA. Because everything feeds into one vSwitch / vNetwork and both sides of the conversation arrive just fine from SwitchB I believe everything is configured correctly on the vSphere server/guest. What could be causing one way packets to arrive on my guest machine from only one interface, but not the other? Could a bad cable be causing the problems from SwitchB?

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  • [SOLVED} How do I restore my audio after uninstalling Ventrilo?

    - by Marcx
    Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. Update: Here's a sample of how I hear others people voices.. Audio Sample After some tests, also the desktop has the same problem. (I tried TeamSpeak3) Here are some details on my laptop and desktop Laptop Dell Studio 1555 Core 2 Duo P8600 2.4Ghz 4Gb Ram Dual Channel Ati HD 4570 512Mb dedicated (up to 2048) IDT High Definition Audio Desktop Motherboard Asus P5KPL-AM Dual Core CPU E5200 2.50Ghz 2x2GB PC6400 Dual Channel Ati Radeon HD 4650 512MB VIA High Definition Audio Both computers have Windows 7 Professional 64Bit. So how do I restore my audio? SOLVED The problem was in router firmware, there was a bug that recognized VoIP traffic as a DOS attack and the router grambled every packet... I've installed the newest firmware and everything is fine :)

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  • Turn old rack server into a telephone server

    - by Jake Elsley
    I have an old server lying around and I am thinking of using it as an internal telephone server. Its main use would be to set up a 1 to 1 telephone system that could be used internally to connect to different users in different offices. I have looked at software like Asterisk, but it seems that this is mainly for external telephone systems. Is this possible to do with Asterisk (or other software) or is this not possible without involving a VOIP company?

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  • Recommended FXO Gateway for UK analogue lines

    - by Bryan
    Can anybody recommend any FXO gateway devices to connect analogue telephone lines to an Asterisk VoIP system. Requirements: Minimum of 4 ports. Enterprise grade - quality is more important than price. For UK analogue telephone lines. - I don't know if this makes a difference or not? I'd also be interested to hear bad experiences, so I can get an idea of which devices to avoid.

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  • Free Software Simulators for SS7, ISDN, SIP, etc., Telecom Protocols.

    - by RBA
    Hi, I am learning Protocols where I have major use of Media Gateway Controllers, Media Gateway, PSTN N/w, VOIP N/w. Calls getting gatewayed from one node to another. Kindly help me in finding out some related software simulators where I can view pictorially the messages being exchanged between the various nodes in telecom architecture. Thanks

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  • How to stop registration attempts on Asterisk

    - by Travesty3
    The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:50] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:55] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:55] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:57] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device <sip:[email protected]>;tag=cb23fe53 [2012-05-29 15:53:57] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device <sip:[email protected]>;tag=cb23fe53 [2012-05-29 15:54:02] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:54:03] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 21:20:36] NOTICE[5578] chan_sip.c: Registration from '"55435217"<sip:[email protected]>' failed for '65.218.221.180' - No matching peer found [2012-05-29 21:20:36] NOTICE[5578] chan_sip.c: Registration from '"1731687005"<sip:[email protected]>' failed for '65.218.221.180' - No matching peer found [2012-05-30 01:18:58] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=dEBcOzUysX [2012-05-30 01:18:58] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=9zUari4Mve [2012-05-30 01:19:00] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=sOYgI1ItQn [2012-05-30 01:19:02] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=2EGLTzZSEi [2012-05-30 01:19:04] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=j0JfZoPcur [2012-05-30 01:19:06] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=Ra0DFDKggt [2012-05-30 01:19:08] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=rR7q7aTHEz [2012-05-30 01:19:10] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=VHUMtOpIvU [2012-05-30 01:19:12] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=JxZUzBnPMW I use Asterisk for an automated phone system. The only thing it does is receives incoming calls and executes a Perl script. No outgoing calls, no incoming calls to an actual phone, no phones registered with Asterisk. It seems like there should be an easy way to block all unauthorized registration attempts, but I have struggled with this for a long time. It seems like there should be a more effective way to prevent these attempts from even getting far enough to reach my Asterisk logs. Some setting I could turn on/off that doesn't allow registration attempts at all or something. Is there any way to do this? Also, am I correct in assuming that the "Registration from ..." messages are likely people attempting to get access to my Asterisk server (probably to make calls on my account)? And what's the difference between those messages and the "Sending fake auth rejection ..." messages? Further detail: I know that the "Registration from ..." lines are intruders attempting to get access to my Asterisk server. With Fail2Ban set up, these IPs are banned after 5 attempts (for some reason, one got 6 attempts, but w/e). But I have no idea what the "Sending fake auth rejection ..." messages mean or how to stop these potential intrusion attempts. As far as I can tell, they have never been successful (haven't seen any weird charges on my bills or anything). Here's what I have done: Set up hardware firewall rules as shown below. Here, xx.xx.xx.xx is the IP address of the server, yy.yy.yy.yy is the IP address of our facility, and aa.aa.aa.aa, bb.bb.bb.bb, and cc.cc.cc.cc are the IP addresses that our VoIP provider uses. Theoretically, ports 10000-20000 should only be accessible by those three IPs.+-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ | Order | Source Ip | Protocol | Direction | Action | Destination Ip | Destination Port | +-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ | 1 | cc.cc.cc.cc/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 2 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 80 | | 3 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2749 | | 4 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 443 | | 5 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 53 | | 6 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1981 | | 7 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1991 | | 8 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2001 | | 9 | yy.yy.yy.yy/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 137-138 | | 10 | yy.yy.yy.yy/255.255.255.255 | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 139 | | 11 | yy.yy.yy.yy/255.255.255.255 | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 445 | | 14 | aa.aa.aa.aa/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 17 | bb.bb.bb.bb/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 18 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1971 | | 19 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2739 | | 20 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1023-1050 | | 21 | any | all | inbound | deny | any on server | 1-65535 | +-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ Set up Fail2Ban. This is sort of working, but it's reactive instead of proactive, and doesn't seem to be blocking everything (like the "Sending fake auth rejection ..." messages). Set up rules in sip.conf to deny all except for my VoIP provider. Here is my sip.conf with almost all commented lines removed (to save space). Notice at the bottom is my attempt to deny all except for my VoIP provider:[general] context=default allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=g726 allow=ulaw allow=alaw allow=g726aal2 allow=adpcm allow=slin allow=lpc10 allow=speex allow=g726 insecure=invite alwaysauthreject=yes ;registertimeout=20 registerattempts=0 register = user:pass:[email protected]:5060/700 [mysipprovider] type=peer username=user fromuser=user secret=pass host=sip.mysipprovider.com fromdomain=sip.mysipprovider.com nat=no ;canreinvite=yes qualify=yes context=inbound-mysipprovider disallow=all allow=ulaw allow=alaw allow=gsm insecure=port,invite deny=0.0.0.0/0.0.0.0 permit=aa.aa.aa.aa/255.255.255.255 permit=bb.bb.bb.bb/255.255.255.255 permit=cc.cc.cc.cc/255.255.255.255

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  • How do I restore my audio after uninstalling Ventrilo?

    - by Marcx
    Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. So how do I restore my audio?

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  • How to take a call without forwarding it

    - by ageis23
    Hi I have a traditional telephony system at home. I also got an voip ATA device connected to the telephone socket. On a normal phone you can just plug it into the wall socket then multiple people on different phones can speak a the same time. Currently for me to take this call whoever answered the call will forward it onto my internal number. Is there not a way I can make it work like the analogue system so I can just pick up the phone?

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  • Connect Cisco SBCS with Nortel BCM (H.323)

    - by D4
    Hi, I am wondering if its possible to connect a Cisco Unified Communications 500 (at a branch office) to Our main Site´s Nortel BCM as a remote Gateway for VOIP Comunication... I´ve had a little experience with branch office telephony but only with Nortel Techonology. don´t know if the Cisco SBCS will play along the other kids. :P thanks in advance... Any thought is more than welcome..

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  • Messenger for voice calls

    - by Rogue
    I have been using Skype for a very long time, but lately I have been facing a lot of issues with the voice clarity. What messenger or VOIP clients that you know out of experience that are better than Skype for voice communication ?

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  • Vlan on astaro 120

    - by Crash893
    (I'm not 100% sure where networking/router questions go this is my best guess) I have a astaro (sophos) white UTM 120 router for work I also have about 11 Voip phones with an externaly hosted pbx (company name = pingtone) Is there any advantage to setting up the phones on a vlan vs making a qos rule that all traffic to my tftp server gets right of way? networking is still a little soft to me Thanks

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  • SIP communicator client

    - by Afro Genius
    I want to build a sip client based on SIP Communicator - the Java VoIP and Instant Messaging client. Basically I need to plug in some how and redirect VoIP to and from my application. Where is a good place to start? If this seems a bit vague, I do apologize.

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  • QoS / PBR Routing Questions

    - by Bernard
    I have a 50Mbs Satellite link and a 10Mbs Microwave link supplying a very remote location. Behind these links, I have a 6,400 seat network - with about 3,000 signed in at any one time. My goal is to send all of the Voip traffic (Google Chat, Magic Jack, Skype, Speakeasy, Vonage, Vonage PC, Yahoo) through the microwave link which has 100ms latency. The rest of the traffic can utilize any remaining bandwidth of the microwave link with excess being diverted to the higher latency (600ms) satellite connection. The problem I've had so far is that most automatic routing configurations weigh the bandwidth heavily for preference - and I'm only wanting latency considered. Additionally, I don't know if this can even be handled with the routing hardware I have at my disposal (Cisco 3640, 3745, & 3845). Any recommendations (or really good starting points) would be greatly appreciated.

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  • Unable to call through asterisk

    - by sk
    I want to create a voip service. I have installed asterisk-1.4 on a dedicated remotely hosted debian lenny distro. I made a sip.conf and extensions.conf so as to place a call between two sip phones(i am using xlite 3.0) installed in some other Windows PC. Whenever i switch this phones the asterisk console shows that Registration from '"1000"<sip:[email protected]>' failed for '122.168.10.254' - Peer is not supposed to register Where xx.xx.xx.xx is the server's IP. i.e my sip phones are unable to register with the asterisk server. Please help me to place call between two sip phones #sip show peers Name/username Host Dyn Nat ACL Port Status 2000 (Unspecified) D 0 Unmonitored 1000 (Unspecified) D 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] # sip show registry Host Username Refresh State Reg.Time # sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 0 active SIP channels

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  • open ports on a cisco/linksys wrt54g2 v1.5

    - by Crash893
    Im upgradeing my router from a netgear fvs318 one of the problems im running into is on the old net gear i have opend ports under the "Add Service " for our voip udp 69 tcp 80 tcp 2000 udp 22026-62025 but on the new linksys/cisco rotuer i don't see any option to just open a port i see port forwarding (which i dont want because i have more than one phone) and port triggering that i have no idea what that does. does anyone know either how to allow open ports on this router or how to use port tiggering to allow my phones to work with the ports listed above?

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  • Make and receive calls from and to PC to mobile and vice versa

    - by Hunt
    I want to route normal phone calls (i.e. calls made from landline or mobile) to VoIP and vice versa. Fr example, if I dial a number from a PC I will be able to call the other person, and the other person is able to see my number on their screen. Similarly, if a person calls me, I can pick up a call on my PC and can see their number on my screen. I don't have any idea how to implement this – how would I go about doing that?

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  • Make own dial-up ISP using broadband connection [on hold]

    - by SkylarMT
    So, I see no reason why this wouldn't be possible. I have a Linux server (a Raspberry Pi to be exact) connected via Ethernet to a broadband ISP. I want to be able to dial a number, have it go through the normal telephone network, onto the Internet via a VoIP provider (I know you can call a Skype user from a landline), to my Raspberry Pi, and then have the Pi connect me to the Internet. I've found guides on making your own ISP, but they all involve a dedicated phone line on the server end. Is there a way to do this with no modem on the server end? I live in an area with a lot of people still on dialup, and if I pull this off I could make some extra money. Thanks!

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  • Running multipul lines through a server.

    - by Kevin Roberson
    I am looking to buy numbers in bulk on DIDx.net. After I purchase the numbers in a particular area code, I want to forward those numbers to other numbers that are outside of that area code. This way it will be seen as a local call versus long distance. I have the customers but I don't have the system I need. I have read about Asterisk, VOIP, SIP, and BYOH. But I have no clue what will be the best system for me. Does anyone have any idea what my next step should be when it comes to hardware and software? Or what type of operating system I should use? I basically want to set up a system like GoogleVoice & Phonebooth.

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