I am trying to connect my SPA122 ATA to my new TP-Link TL-WDR3600 wireless router. This setup worked with my old router (NetGear WGR614) but I cannot seem to get it to work with the new router.
I am connected to the Internet through a Huawei MT130U cable modem with nominal 28MBS bandwidth. I have been told that the throughput of the router in the SPA122 would restrict that so I had connected it to my old WLAN router in the DMZ. This worked fine before but not with new router.
I have a cisco RV220W router used in a home environment. Recently I purchased a Siemens Gigaset A 580 IP phone. It works OK so far but I would like to optimize the configuration for VoIP traffic and apply QoS for VoIP on this router. Any guide with the recommended configuration and preferred settings of the same? I have not used QoS previously
I am unable to get any configuration parameters to work on my SRP547 for VOIP. ADSL2 works fine. Equipment replaces a ya.com (Arcadyan ARV4518PW) which worked ine albeit with significantly less features. Problems encountered due to the lack of equivalents for some parameters and an overabundance in the new unit which simply don't exist in the Arcadyan product.
I have an SRP547W hooked up as the office router with the standard office phones connected via the telephone ports at the back of the unit using 2 SIP lines as well as the PSTN by dialling hash first. We have just added a new staff member and bought an SPA303 with the intention of connecting it through registered SIP lines on the SRP547W, and hopefully have the facility to use the PSTN line when the SIP lines are busy.
The problem is, it connects to VLAN100 and gets its IP address and initializes fine however no lines show as configured and it can't make or receive calls. What do I need to configure on the SPA303 to tell it to use the SRP547W as its SIP Server/Proxy (not sure of the terminology).
I have a RV042 with a DSL (WAN1) and cable (WAN2) internet connection in Load Balance Mode. The DSL provider also provides internet telephony when registered via his line. When I disable the WAN2 port, my IP phone successully registers with the registration server of the DSL provider. I also defined protocol bindings for SIP (port 5060) and RTP (ports 5004 to 5020) to be bound to WAN1. My IP phone is set up to listen on only these ports. [code] With these protocol bindings in place when I re-enable WAN2, then after some time the phone reports "registration failed".Do I need to set something else apart from protocol binding to force the VoIP traffic to go via WAN1?
We have a RV082 configured with two ISP Wan connections. We recently implemented a VOIP phone system (SIP) (192.168.1.50) that is being used in appliance mode on our network. We currently have two WAN connections Load Balanced. My goal is to configure all my VOIP traffic to go out through the 1st ISP and the rest of the data through ISP #2. is this possible to achieve using the RV082? We are using a Skype SIP Trunk connection.
We have recently purchased an RV082 router for our company. I have successfully managed to establish a VPN connection through PPTP and through Quick VPN (IPSEC?)
When i am connected through PPTP my software SIP client has 0.5% packet loss which is totally acceptable. When i am connected through Quick VPN my software SIP client has a 50-70% packet loss.
i have done the following settings on the router: Bandwidth Management:
High Priority for upstream and downstream UDP 10001-20000 and UDP 5060-5080
Wan Connection settings: MTU: 1492 bytes
I would like to know if there are any other settings that will improve VOIP over Quick VPN since RV082 supports 100 connections compared to 5 for PPTP.
Advanced networking but have managed to get our small office IP PBX running over a SIP Trunk. The only real problem we are having is choppy outgoing audio when there is other heavy outgoing traffic on the network.
My understanding is that I need to set some QoS parameters, which I have played with but it didn't seem to be useful. I mostly dealt with allocating bandwidth. I now think I need to somehow prioritize the outgoing RTP packets from our PBX (which runs on a PC on our LAN) to avoid the choppy audio. My research shows this can maybe be done with something called DSCP 46 and my router does support that -- how to exactly set the configuration.
I have the router for over two years, Present Firmware is V2.0.1.3, I have Polycom phone connected to the router using 8 x 8 VOIP service. The phone was working fine untill yesterday. 8 x 8 tech support suggested getting a different router. I have the same router at home and tried the same phone on my home router - it worked. So, the problem is not the phone but int he router. I tried to call Cisco tech support but found out that the warranty support expired and I need to buy a support plan. Called to buy the support, and gave up after an hours on hold. Left message.
I have VOIP and was provided a modem, a Tilgen Router (Vood 452_A). I also currently have a Linksys WRT54G router. I also have a home surveillance system with a DVR that can be accessed via the net. I am trying to allow the ports to open so that traffic can navigate down to the DVR.My IPs:My VOIP has provided me with 6 static IP addresses. For sake of this issue they are:
67.xx.xxx.153, through .158My subnetmask is set to:
255.255.255.248My Settings:
The Modem is connected to world. To the modem I have the Tilgen Router connected. The Tilgen is set with an ip of .153 and the subnetmask listed above. DHCP is turned on and has IP range set for .154-.158. It is set in Bridge Mode with NAT and Firewall turned off. To that I have the WRTG54G router connected and assigned the router an Static IP of .154.. The cable is going into the Linksys WAN port.From a computer connected to a LAN port on the WRT54G, I see my public IP as .154, currently my internal IP is 192.168.1.15 so I believe I have to assign one of my Static IPs to this PC, however when I try to connect to the Linksys Router (.154) from the outside web, I am get a time out error. (also from a ping test.) My thoughts are to assign the Linksys to DHCP and use a range of .155-.158 to give each lan item a static IP. My question is will this work? Will each item in the network have its own IP and if I forward the port to the DVR that i needed, ie 80 for web client or 6100 for desktop client, will that forward through?Or do I set it to Router Mode and use the Tilgen to assign the ips.
On the Tilgen I also have an option for a DHCP relay or to turn off DHCP.
For clarification here is the setup chain Broadband/DSL Line>>Tilgen Modem>>(LANPort1)Tilgen Router(LAN Port2)>>(WAN) Linksys(LANPort1)>>DVR/Computers Found this walkthrough on setting the WRT54G into a Switch. If I did this, in theory the VOOD would assign those computers/dvr connected downstream via the WRT54G a static ip from my range set on the router. Then in theory I would not need to port forward port 80 since the DVR would be on its own public IP.?
I'm going to move offices into a shared situation with 3 companies. Each company will want its own private network so there's no snooping between companies. I am planning on using VOIP for the phone system (Nextiva cloud based). Is it possible to set up the system so that each company has access to the VOIP system but yet remains sequestered in the their own network for everything else. I was hoping to do this with one data port at each workstation using Cisco SPA-303 phones. The way I understand this, is that the phone plugs in to the data port and you daisy chain the workstation off from each phone. Is this possible to do this while having the system I described? Another wrinkle is that I'd also like all the networks to be access shared printers.
I have 4 remote sites that are using a ASA as thir firewall / router. I'm setting up a full mesh VPN between all the sites. One of the sites have a UC500 and the other sites access that UC over the VPN tunnels. I would like to set up some basic QoS for the VOIP traffic
The site that has the UC will have multiple vpn tunnles coming in from the remote sites. How will I do QoS with voice traffic on that site?
I have a Cisco ASA 5540 running 8.2(5). When I dial a phone on the other of the the VPN the first time I get a blank after it rings(i.e when the voice mail get activated if someone picks the phone up), however works the second and consequent times i dial.
A little background. Two sites A and B connected via IPsec Tunnel. No problems in communication except for the VoIP issue. A Phone in on site A(172.17.168.x) and other on site B(192.168.103.x). Site A and Site B is connected via an IPsec tunnel on the Cisco ASA. First call fails. Second call works. Result of a packet trace is also the same. The UDP packet get drops when tried for the first time but subsequent ones pass.
How do i connect two internet routers to one IP address range?I currently have one broadband router connected on 172.16.0.3 subnet 255.255.0.0, I have a second line which is a copper ethernet EFM 2Mb line connected using a Cisco 1800 router configured on 172.16.0.254 subnet 255.255.255.0, no DHCP is configured on either router. the dhcp is provided by a windows 2003 server on our LAN which is also configured on the same LAN IP range. if i connect the Cisco router into the same switch as the broadband router my network devices stop working and dont respond.
If i check the broadband router it shows all the connections to it as having the same MAC address as the Cisco router?? if i disconnect the Cisco router everything returns back to normal and works perfectly.What I want to achieve is to be able to run both routers on the same LAN to provide internet access by configuring the pc's to route through a specific router. the DHCP I want to stay routed through my original broadband line. at the same time i want all pc's to access to the lan devices (ie. servers and printers) what the correct configuration would be for the two internet routers would be?
I am trying to connect 2 wireless routers to expand my wireless signal. I want to connect the router 1 (in 2d floor) to a powerline av and then get the router 2 to the powerline (in the ground floor) so i can have wireless there too. I've tried some configurations i found online, but can't get to make the network functioning. The routers and powerline av are all dlink. I don't mind having 2 networks, but it would be better to have only one
I have a laptop connected to a net gear wireless router (for internet), but I am very close to buying a Belkin network USB hub to make my hard drive wireless, but this requires a router. I do not want move my hard drive to where my net gear router is, so I need to get another router, i am looking at a Belkin router so that I can make my hard drive wireless (This is the only reason for the router, it will not be used for internet) so I can keep my hard drive in the same place.
having some issues. My basic VOIP network I can get to work no problem uner Vlan 1. But when I try tomake multiple basic networks to connect and put them in to diffrent Vlans such as Vlan 2, 3, 4 and conect them the phones now say configuering IP.
We just purchased cisco 2960 for our VoIP needs and we are using polycom phones, and Phone and Computer will use same port. Since Polycom phones are capable working with CDP protocol and we are hoping to get another switch to expand VoIP network. I found easiest way of setting up each port is as following (from the cisco tutorial)
My first question,when we are using switchport voice vlan dot1p ,I thought we instruct the switch port to use 802.1P priority tagging for voice traffic and to use the default native VLAN (VLAN 0) to carry all traffic.Do I still need to create a Vlan 20 for data and Vlan 10 for voice ?
Secondly,same tutorial adds these commands as well,Do you think for our set up, using those commands are feasible ?
Thirdly,when we get another switch and do the same configuration for the second switch, can I use any port on Switch 1 as uplink without doing any configuration ?
I have a little weird request at work. One of our offices would like to split the VOIP traffic. At that office we have a 10MB primary and 3MB backup circuit. Currently the phones are routing over the 10MB circuit. The General Manager would like to use the 3MB backup circuit for VOIP traffic. For the 3MB we have two T1 lines bundled together in a multilink. Configuration is bellow if needed
I have a 887 setup as a EasyVPN server, and a 861 as an EasyVPN remote - network extension mode with split tunnelling.This works fine - I can ping and connect to machines across the tunnel.However if I setup a VOIP handset to connect across the tunnel it registers and calls work, but drop after 30secs....I know this is normally a firewall or nat problem, are easyvpns firewalled or natted?
I am fairly new to Cisco, but am trying to configure a 1921 router to give higher priority to SIP/VoIP traffic (Port 5060) than everything else.The connection is only 4Mb and is getting hit hard by video streaming, I don't want to block this, just make a lower priority.Any ideas where I am going wrong?My current config is as below.The IP addresses have been changed for security reasons, but in reality are both in the same range, i.e. are both external IPs, so I am not sure if this is causing the problem. Do I need NAT for QoS to work?
I am tasked to connect my VoIP phones from remote site to my corp site. Basically all remote phones will be registering into a VoIP server in corp site. I have a site to site vpn tunnel established already from remote site to corp site. My hardware includes the following:
-Cisco ASA 5505 -Cisco small business POE switch SF300 24p -Avaya 2015p VoIP phones
Successfully Register remote VoIP phones to corporate VoIP server 10.30.18.55.I have already configured vlan1 10.30.15.0/24(inside lan) and vlan2 public int(outside Internet) which my dmz only allows 2 per my basic asa licensing.When I connect my phones and register it states "subnet conflict" unable to register.
I am trying to get QoS for my VoIP system setup on several SGE2000p switches and have got a question...How do I define the ACL for RTP? As far as I can tell it will not let me enter a UDP port range for the RTP traffic... And I cant imagine creating rules for each port would be very effective either. So, how can I define an ACL to cover the RTP traffic so I can classify it?
I am encountering delays when making any calls using VOIP. I understand that there is always a delay using VOIP but not as much. I've done a bandwidth test on my network and everything seems to be Ok, we have 4 T1s with 55 Reps using it. There is nothing choking up the network.
Trying to select a router that will work well networking a computer with a nettalk duo, on a limited bandwidth connection: .3/3Mbps up/down. From what little I've been able to find on this, QoS bandwidth control seems critical, yet the list of recommended routers from Nettalk seems to favor routers that don't have this feature. On the other hand, the list apparently hasn't been updated in a year and a half.
