How to run Cisco call attendant running in the background.I have a user (who’s giving me a headache) that wants the Call attendant software to not pop-up when a call come in.She doesnt want to lose her mouse cursor when call comes in.I have looked at the call attendant user guide and not able to find this feature.This user says that the previous Network admin did something to run it in the background.
View 1 RepliesI got a quad core i5 750 and pci-e intel pro CT.It seems when I leave a game running in a window (is an old game, final fantasy 7) although the cpu is nowhere near full load in fact barely quarter load I am getting jitter on my lan traffic. Took me a while to diagnose as I didnt think that would be cause. It happens whether hardware offload is on or off on the card. This also didnt happen with the same network card in my core2duo machine, so even tho that had less power it didnt have a problem with this usage.
Is this a possibility of a saturated DMI channel on the P55 chipset, sending data for the game at same time as nic data? or probably something else?
I have a standard ADSL modem which connects to the internet. On the inside I have a few computers within my LAN.when the modem receives an incoming request from the internet for a connection to one of my LAN computers e.g. a Skype incoming call, how does the modem know which port to forward that traffic to on my internal LAN? i.e. how does the modem know which of my computers is running the skype application that will answer the incoming call? I know port forwarding normally handles this sort of thing, but in my case, I am not using any configured port forwarding rules so how does the modem know where to forward skype traffic?
View 2 Replies View RelatedI have a router SRP521W I wonder if you can setup a self-attendant or ivr to this device.
View 4 Replies View RelatedEveryday 04:00 AM, I set repeatly configuration sync. But, Since 04/16/11 04:00:00, Background tsak is not working. So, I was wcs stop -> start (04/18/11 08:30) this situation is randomly occurrenced. I found similar symptom, bug CSCtf23192. This bug fixed 6.0.202. But, I used 7.0.164. I want to know that this bug occurence 7.x and fix in 7.0.172.
04/16/11 03:55:37.774 INFO [monitor] [PollSerializationLock-1] SiAqStatspostNetwork Ending updateServiceDomainNodes for Area 165.243.138.6-12603442904884/GSIDC_12F, elapsed = 0 04/16/11 03:55:37.837 INFO [monitor] [PollSerializationLock-1] SiAqStatspostNetwork Ending updateServiceDomainNodes for Area 165.243.138.6-12603443248315/GSIDC_13F, elapsed = 0 04/16/11 03:55:37.837 INFO [monitor] (code)
i want to know the background process of computer networks
View 1 Replies View RelatedI have 30 switched in my corporate network it’s all up and running all switches running by default configuration and connected to WS-C4506 core switch our dhcp server pooling 192.168.100.1/27 network. Now we need to configure new Vlan for finance department this department has more than 200 users. If my server distributes 192.168.200.0 range ip can vlan2 automatically assign ip 200.0 addresses to finance department.All switches running default config no ip address assigned.
View 9 Replies View RelatedI need to replace an existing ASA 5540 with a new ASA 5525X. I would like to pre-stage and configure the new box with the existing config, migrate license and export certificate files before swapping it with the old one during a change window. The new firewall will run 9.1 on deployment. Now the same 7.2(4) cannot just be copied over to 5525X running the minimum 8.6 version. There is a Web based tool available at [URL] according to Cisco documentation but the page does not load for me (Cisco intranet only tool ?). Is there another tool for automatic conversion ?
View 3 Replies View RelatedI have a telephony system already in place with 6 FXO active ports configured on a 2821 Router.The thing is that I am not able to make an ip phone forward calls to a mobile No.I can make calls to the same mobile number when dialed from the phone but the call routing seems not to function correctly when call-forward all is configured on the ephone-dn.I would like to focus only on voice-port 0/0/0 and 0/0/1?
voice-port 0/0/0
supervisory disconnect anytone
output attenuation -3
echo-cancel coverage 32
compand-type a-law
cptone GB
[code]....
we are running callManager 7.1 (user accounts are tied to AD)... one night our call manager got rebooted by IT. upon reboot, CM failed over to the Subscriber node. Some how IT got Callmanger back over to the Publisher node.....No now of us can log into CallManager using our AD accounts...
View 2 Replies View RelatedI 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 use my sim card in modem connected to pc to surf. is it possible to receive calls as i surf through this sim card?
View 1 Replies View RelatedI would like to use the Cisco ATA 186 as a SIP device with Cisco Call Manager 7. However, I have found that the SIP firmware for the Cisco ATA 186 is unsupported by Call Manager 7.Therefore, In order to use the ATA as a SIP device with the Call Manager, I believe I will need setup the ATA as a 3rd Party SIP device by doing the following:
1. Disable the TFTP option within the ATA, so that the SIP firmware is not overwritten by firmware from the TFTP server.
2. Install the 3.2.1 SIP firmware on the ATA, since the ATA is currently running the 3.2.4 SCCP firmware.
3. Register the ATA, as a third-party SIP device, with the Call Manager .Should I register the ATA as a 3rd party device?
Is it possible for a 9971 and ipad2 to have a video call either with Webex or another app? We're getting both soon, just seeing if anyone has tested it yet.
View 3 Replies View RelatedI a using a Ciscoet 2800 Series router with a Call Manager 5.0.1. When I make a call I g this error:
*Dec 12 20:30:38.575 Potential Mute Call:
*Dec 12 20:30:38.575 Call ID1=3 Call ID2=4 ConfID=2
*Dec 12 20:30:38 575 Leg1: CallID=3; TX Packets 398; RX Packets 400;
*Dec 12 20:30:38 565 Leg 2: CallID=4 TX Packets 398; RX Packets 400;
I can be heard but do not here anything. No ringback at all.
I have installed LMS 4.1 and discover all the devices (router/switches) but i want to show IP Phone on the LMS. I am unable to discover call manager in LMS 4.1 topology.
In Call Manager 8.6 in Cisco Serviceability under snmp setting i have enable the read community string and check snmp, MIB service are running.AM i using the correct proecdure. How can i get the Call Manager on LMS server so that i can see IP phones on LMS topology. 8.6 is installed on VM Ware.
I setup one network where LMS is in subnet 192.168.5.0/24 and CUCM is in subnet 192.168.1.0/24. both are reachabe to each other and both are also in different EVN/VRF. when i try to discover the CUCM from LMS 4.1. It discover only routers and switches. I am unable to find any CUCM 8.6 server, i did the snmp read community setting in call manager under serviceability.
In addition, IP phones are appeared as END host not as IP Phones. (find attached image)
We have installed a SRST in one of our remote offices. They have ISDN Primary, and we have a 6mb MPLS link to our main HQ with 400k of QoS on the line. During the day, both Internal calls over the MPLS and external calls over the ISDN break up, but it doesnt happen all the time, i.e in a day the morning will be ok, then the users experience the break ups in the afternoon, then its everything is ok again.
The only factor that the calls share is the Lan infrastructure i.e the 3750G switches (7) of them which are stacked.We did have MAC flapping errors, and our telecoms provider suggested that it maybe these causing the issues, however we have got rid of all MAC flap errors, and are still receiving voice break ups.
Because the Lan switches are the only "item" both calls share, I am wondering now if the setup of the switches is correct. [code]
In Age of Empires ESO will let me log in to game, but fail to join games in lobby
View 1 Replies View RelatedHow to find the no of the incoming call by at commands.
View 1 Replies View RelatedI have a 2811 with CME 8.5; I recently added a SIP trunk and can make & receive calls just fine. I noticed today that when connecting to an external IVR system, I can't send additional tones (press 1 for Sales, for example). I have two DIDs, same provider, both doing the same thing. The config for one of my SIP trunk dial peer and translations are shown here - I think this is what's appropriate to review but can add anything else that's necessary. I've also omitted some dial peers for X11 and international dialing for brevity.
View 2 Replies View Relatedcreate a hunt group for the people on call. I have plane this below configuration so far, and I would like to associate the cisco ip phone number with the personal mobile.
1) how can I associate the office number with the mobile one?
2)how and where I set up the forward no answer?
1) Create a line group, and add lines according to the user that we have in the group. Set the Distribution Algorithm to Circular.[URL] I need to create 1 line group per each user. Step to create a line
a)Choose Call Routing > Route/Hunt > Line Group
b)To add a new line group, click the Add New button
c)n the Line Group Configuration window that displays, enter a name in the Line Group Name field. The name can contain up to 50 alphanumeric characters and can contain any combination of spaces,
periods (.), hyphens (-), and underscore characters (_). Ensure that each line group name is unique to the route plan.
d)Choose the appropriate settings as described in Table 36-1. Table link ( http://www.cisco.com/en/US/docs/voice_i ... #wp1053560 )
e)To add or update this line group, click Save.
2) Create a hunt list, and add the line group Step to create a hunt list
a)Choose Call Routing > Route/Hunt > Hunt List.
b)Click Add new
c)In the Hunt List Name field, enter a name. The name can comprise up to 50 alphanumeric characters and can contain any combination of spaces, periods (.), hyphens (-), and underscore characters (_). Ensure each hunt list name is unique to the route plan.
d)In the Hunt List Name field, enter a name. The name can comprise up to 50 alphanumeric characters and can contain any combination of spaces, periods (.), hyphens (-), and underscore characters (_). Ensure each hunt list name is unique to the route plan.
e)To add this hunt list, click save
f)The system checks the Enable this Hunt List check box by default for the new hunt list.
g)Add at least one line group to the new hunt list.Adding Line Groups to a Hunt List (Associate a line group to hunt group)
a)Choose Call Routing > Route/Hunt > Hunt List.
b)Locate the hunt list to which you want to add a line group
c)To add a line group, click Add Line Group.The Hunt List Detail Configuration window displays.
d)From the Line Group drop-down list box, choose a line group to add to the hunt list.
e)To add the line group, click Save.The line group name displays in the Hunt List Details list on the left side of the window.
f)To add more line groups to this list, click Add Line Group and repeat Step c through Step e.
h)Click save and then to click reset to reset the hunt list. When the popup windows display, click OK.
I have CUCM 8.5 in a lab environment that connects to a Cisco 2801 which has a VIC2 FXO card on board, I have believe that I have configured the MGCP correctly and also configured my CUCM correctly but I'm just getting a fast busy when I make a call through the PSTN, I have tried removing the Ethernet cable from the Router just to test what happens when I dial external and again I get fast bust.
Here is my configuration setting: -
From CUCM gateway details:
Gateway Name MGCP
Protocol MGCP
AALN/S0/SU1/0@MGCP
AALN/S0/SU1/1@MGCP
AALN/S0/SU1/2@MGCP
AALN/S0/SU1/3@MGCP
[code]....
regarding device package for Cisco MCS 7835-I3 [Cisco 7800 Series Media Convergence Servers] . I have added Call manager in LMS 3.2.1 but showing ? as there is no device package installed in LMS.
View 2 Replies View RelatedCisco IOS Software, 1841 Software (C1841-ADVENTERPRISEK9-M), Version 15.1(4)M2, RELEASE SOFTWARE (fc1)
When the client on a Branch Office LAN realizes calls to skype, there is a sharp loading of CPU on Branch router Cisco1841 (all Inet traffic goes through DMVPN-Tunnel Interface to proxy on Central Office).
[code]....
I have a client using a VOIP service to a third party provider (RingCentral). They are connected via Cable ISP (6mb) to the Internet and now experiencing performance issues with their VOIP service. They indicated that the call can be heard but that there is jitter and choppines in the call and they have to place a regular landline call. Their provider recommended using QOS to improve. I did not see anything straight forward on the ASDM interface to do this and figure it may require command line to accomplish.
They have Cisco IP 303 and 5252G2 phones which connect through an ASA5505 7.2(4) to their provider for service. Apparently the voip app uses the following ports:
UDP
5060-5090
8000-8200
16384-16482
What would be the best solution to improve performance or perhaps traffic shape / priortize traffic to work. I assume this may be happening if there are heavy downloads or activity happening on the network. The ASA5505 is on 7.2(4). Some coded examples for the above info.
I have seen that when ever i am using the packet data connection (GPRS/EDGE) on mobile and if a Voice Call comes, the Package data will be stopped and once the call gets disconnected it will resume sending the data.For 3G Data the same is not happening, at the time of voice call it will continue sending the Packet data.
View 1 Replies View Relatedone last try before boxing up my new DIR-655 and taking it back to Microcenter. I've opened a case via email and got the standard response, below.I have a Dlink DP-300U print server that works fine with my 'old' Belkin router, a F5D-7230-4 wireless unit. Nothing wrong with the Belkin, I'm just getting some service drops from my lousy AT&T ifitl 1.5mbps ISP and thought a new router might work. I actually read the manual (!!) before installing the DIR-655 and even followed it. The print server just isn't visible on the network. Everything else is, though. The 'suggestions' from Dlink tech support below are very useful. If I could 'see' the print server to verify its IP address, we wouldn't be having this conversation. I've rebooted everything, several times and even reinstalled my old Belkin router (which always sees the print server.) And Dlink, no...I'm not interested in paying you $32.95 for up to a half hour to maybe fix a problem caused by your new device. And dropping my call after almost 15 minutes on hold is not cool, either. With reference to the issue you are facing, we suggest you to kindly ensure that the IP address assigned to DP-300U print server is in the same network range as that of your DIR-655.
NOTE: The default IP address of DIR-655 is 192.168.0.1 Once it is done, try to ping the IP address of DP-300U using any of the wired / wireless client connected to DIR-655.note the DP-300U is not currently supported in North America. D-Link offers a premium fee based support line that will be able to support any issues you have with this product.
The WRP-400 router has two phone ports - does it support direct calls between them?
View 1 Replies View RelatedWireless Phone --> 881W router --> VPN --> ASA5510 --> LAN with CCM and Voice Gateway
Phone registers to CCM Internal calls work External calls connect but get no voice in either direction
What is the reason to call a router as data link layer device
View 1 Replies View Relatedi followed the instructions on the link and I can not for the life of me get my NAT to display OPEN or even Moderate. Always says STRICT. I have tried all the link and all the videos about forwarding ports followed all the instructions to the T. But the PS3 in the DMZ. And still, Nothing. It's BS.
Using router FW: 2.00NA
Ports Forwarded: TCP 80-81,443,5223
UDP 2005,3074,3075
TCP 3074
UDP 3478
Assigned QOS engine rule to the IP of my PlayStation and set it to priority 1.UDP Endport Filtering and TCP Endport Filtering Both set to independent?Enabled Anti-Spoof Checking?
Do you know if it is possible to filter TOIP flows between call server (Siemens technology) and phones ?Specialy, PIX is able to support dynamic ports opening?? Is there an ALG embeded?Is it required to upgrade PIX or not? is required a special licence??
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