Cisco :: Is There Any VOIP Laps For Packet Tracker
Nov 16, 2011As the thread title says, I just want to gain better understanding by practising the config.
View 1 RepliesAs the thread title says, I just want to gain better understanding by practising the config.
View 1 Replieshaving 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.
View 1 Replies View RelatedWe have a customer running a Asterisk based Phone system, using Snom 320 phones and a ESW-520-24P as their switch.
Recently we have been confronted with an issue where every 35 seconds their is a very obvious drop of audio for both parties of a call. We know it is not the phone system as we have run tests directly over the BT copper lines that have no drops what so ever. Unfrotunately when we introduce the LAN in any way, we get the drops every 35 seconds.
We are running the whole solution on 1 vlan, yes I know probably better to use two, but the Cisco cannot automatically detect the Snom 320 phones to swap it to another VLAN, which complicates using a VLAN setup.
I really need to work out how to stop these packet drops, but I am new to the world of Cisco and dont really know where to look or what to try.
How do I fix packet loss on VOIP? Teamspeak, Skype, MSN Voice all have inbound jittery clipped robotic voice. Teamspeaks shows "total" packet lost, climbing to at least 20% with Speech packet loss 50-65% then I lose connection altogether. All other network programs (Firefox, streaming music, videos, online gaming) work fine.I have ATT DSL, using a Netgear Wireless G Router w/ built in Modem (model DG834G). The network has 1 attached device, my Dell laptop (which is usually powered off).The packet loss problem is identical on my desktop pc and laptop (both using the same network). Problem exists when nothing else is running, only Teamspeak (or another VOIP) I have replaced all wires, cables, filters. Power cycled modem. Problem exits with or without Port Forward.*running Windows 7 {64 bit}, AMD Phenom 8450 Triple Core, 2.11ghz [3gb RAM] Asus M3A78 Pro Motherboard
View 3 Replies View RelatedI have followed all instructions to forward port 22 on my netgear router WNDR 3400v2. It says it forwarded but according to port tracker. Its closed. I am having this with two different locations. All IP addresses are correct and the router can ping them. It just wont open. Using SSH program for port 22 access.
View 10 Replies View RelatedI would like to configure IP SLA with ICMP tracker. What is the minimum IOS & Feature required in cisco 1800 Router?
View 2 Replies View RelatedWe have a mix of 1231 and 1242 access points in an LWAPP environment. They are connected to WiSM controllers in our 6509's and are managed by a central WLC. Everything is running version 7.0.230. My question is can I apply MAC address filtering on a select group of AP's to restrict access to a specific SSID broadcast on these AP's without affecting other AP's also connected to the same WiSM?
View 3 Replies View RelatedWe have a 5508 controller with about 40 LAPs, running AIR-CT5500-K9-7-2-103-0-FUS software.The need has arose to have LAPs in various locations that only broadcast specific SSIDs.
I've got seperate dyanmic interfaces already created.We have 3 "Normal" SSIDs that we broadcast to the LAPs already associated with the 5508.2 more SSIDs were created (2 different purposes).
I just can't figure out how to get it to certain LAPs to only broadcast the specific SSIDs, while the rest broadcast the normal 3, but not those 2 new ones.
I am at one of our remote offices and I am noticing my laptop, despite excellent signal strength is periodically losing IP connectivity on the wireless network. When it drops, all of my IP connectivity stops (pings fail, RDP sessions "await reconnection", etc...). The lower right corner still shows I'm connected to the hidden WPA2 Enterprise SSID. The only way to reconnect is to select disconnect on it, then click connect again. Immediately everything IP based starts working.
There is a 5508 controller in the headquarters. The site I am at has a 30mbps fiber point to point WAN to the headquarters. This site has 2 x 3300 series LAPs which are very good coverage. H-REAP mode is on so traffic terminates at the local office because it is more efficient than traversing the LAN twice for things like local file and print sharing, dhcp, proper active directory sites and services mapping, etc.
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.
First time
ASA5520# packet-tracer input inside udp 172.17.168.95 10000 192.168.3.103 10000
Phase: 1
Type: ROUTE-LOOKUP
Subtype: input
Result: ALLOW
Config:
Additional Information:
in 0.0.0.0 0.0.0.0 outside
[code].......
How can I do a VoIP Install/Repair?
View 3 Replies View RelatedWe 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)
Switch#configure terminal
Switch(config)#mls qos
Switch(config)#interface fastethernet 0/1
Switch(config-if)#mls qos trust cos
Switch(config-if)#switchport voice vlan dot1p
Switch(config-if)#switchport voice vlan 10
Switch(config-if)#switchport mode access
Switch(config-if)#switchport access vlan 20
Switch(config-if)#exit
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 ?
Switch(config-if)#switchport priority extend trust
Switch(config-if)#priority-queue out
Switch(config-if)#spanning-tree portfast
Switch(config-if)#spanning-tree bpduguard enable
Switch(config-if)#exit
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 ?
my config and all the show's ive run sofar tryign to figure this out, but the policy map isnt matching the traffic for some reason
View 9 Replies View RelatedI 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
3MB Circuit
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?
View 9 Replies View RelatedI 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?
View 6 Replies View RelatedI 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?
View 4 Replies View RelatedI 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.
View 3 Replies View RelatedTrying 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.
View 5 Replies View RelatedRecently at my company we have started using Polycom i335 IP Phones from 8x8 inc, the thing is that while trying to use it directly through a switch it does not register with the server and I get a "url call is disabled" msg on the LCD of the Polycom. On that same switch I am using an Aastra 6755i IP phone, it is working absolutely fine, although it is using the same technology and ports.Now if I connect my laptop via wireless or if I connect a PC through wire and use ICS to share the internet and connect the Polycom to the shared Ethernet port, the Polycom works fine! On the same network Vonage works fine as well. but this Polycom only and only works via ICS?
View 1 Replies View RelatedHow can I connect two laptops via VOIP?
View 1 Replies View RelatedAny VoIP box which has from 3 to 10 phone ports?I'm sourcing the box for a business that wants to connect about 8 phones to the box, then have the calls terminated on our partitioned platform hosted by a third party.
View 3 Replies View RelatedI have a Voip application on my Windows Vista Home Premium. It is called Reliance iCall. I was not able to call from this application for a few months and so i thought there was something wrong with it. But just yesterday i thought of connecting my modem directly to the desktop and the call went through without any problem. It is only when i connect modem to computer through router that this application does not does not do anything.
What could be in this router that is blocking the calls from making a connection? There is also no setting to configure in this VoIP application. I dont know what to do. And also dont want to keep on switching modem's ethernet cable from router to computer and back.
I'm looking for a VoIP solution for one of our remote sites.I'd like to keep it self contained and not attached to our current Cisco Call Manager.I had a couple sites running Asterisk from EvolutionPBX (URL), but it appears they've gone out of business.I've come across many different vendors for a solution, but of course they all claim to be the best. I'm just curious what others are running.What I really liked about Evolution PBX was how easy it was to setup.I basically need about 25 to 30 extensions, after hours voicemail, and a hunt group.
View 5 Replies View RelatedIs it possible to connect an analog phone to an FXS port on a CME router and a VoIP phone to a switch connected to said router and have voice connectivity between the phones? Also, is it possible to connect an FXO port on that same CME to a RJ-11 wall jack to connect to the PSTN and be able to call that VoIP phone as well as the analog phone from my cellphone? I'm trying to tie as I read the CCNA Voice OCG.
View 5 Replies View RelatedI 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
View 1 Replies View RelatedI have a Cisco 877 on an ADSL connection. QoS isn't doing the trick -- I need to reserve 200 meg or so of my outbound (upstream) bandwidth for VoIP to end complaints about voice quality. Any example of how to classify SIP, RTP, IAX, and Skype traffic and put a rate limit on anything that doesn't fall into that category? The VoIP phones also are in their own IP range on the LAN side if that would make things easier...or I could even connect them into a specific port on the internal switch in the router.
View 9 Replies View RelatedI want to implement VoIP on my client network, currently the data network is using SG300-28P small business switches for user access. According to my design the IP Phones (Cisco 9971 and 7942G IP Phones) are to connect to the small business access switches while the user PCs connects to the IP Phones.
My concern is I really don't understand how the small busness switches will advertise the voice VLAN to the IP Phones. I understand that the switches are suppouse to use LLDP/CDP for this but it seems the model I have can only do LLDP. The IP Phones and the PCs connected to them will be recieving IP addresses from an uplink L3 managed switch.
how LLDP works (particularly regarding this scenario)? Does it matter if the small business switch is in L2 or L3 mode for the VoIP implementation?
We have to set up voip for our network(for 50 phones not he cisco phones).
I need to just the route the voip traffic to gateway address of telephonic company(1.1.5.7) where they provide us the connectivity for the setination call.
What sort of protocols should i have to enable in pix i saw the concepts like sip, h323, ras, skinny.
We are using only voip for asa and no data or other traffic should be allowed.
inside adrees: 10.10.10.0/24 for all voip phones
outside:121.21.22.1
telephoneic gateway: 1.1.5.7
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.
View 7 Replies View RelatedI have configured 2 Vlans on the network. 1 - native Untagged for data traffic and 100 Tagged for VoIP.I have 4 SGE2010p switches 2- of them in stack working in L3 mode and 2 connected to the stack via single ethernet links in L2 mode.I have IP phones with trunked ports -Vlan100 tagged and 1 untagged. I have set CoS 5 for Voice Vlan 100 on the phones and CoS 5 to queue 3 mapping (in basic mode) on all switches.But it seems to not work at all.Should I use Advanced QoS mode?Where Can I find some QoS configuration example for Voice Vlan on SGE2010p switches?
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