Search results for query: *

One thing that comes to mind is that you should not have to do packet captures just to see if you get caller ID. One thing you can do is actually look at the voice gateway that have the voice E1s. Do a "debug isdn q931" or "debug voice ccapi inout". Watch as the phone call comes in. If it is...

Absolutely this can be done and has been done many times before. On the CCM side, you will need a route pattern to send the call to an Inter-cluster trunk or as an H323 Gateway(CME's IP address). From the perspective of the CME router, create a voip dial-peer that will send patterns that the CCM...

I had the same problem with Unity. It was not a new integration. Half of my ports lost registration and resetting them would not register them. A reboot of Unity fixed the problem.

I have the same setup like you, but with a 2621XM. My provider was Cox Cable, and they remember my mac address from my old linksys router. I simply turned off my modem for 10 minutes. After 10 minutes, I turned the modem on and plugged in the 2621XM, and I was given an address. This could be...

One thing you can do a Erase Configuration on the IP Phone. It is option 33 under Network Configuration. You can ensure DHCP is Enabled in here also. Sometimes IP phones still remembers factory ip addresses/settings. This has fixed things for me before. Next is to check the switchport...

Just setup the VG using MGCP. Since you will want SRST, you will have to configure it and put in dial-peers for it. Works perfectly fine for me. Or just go H323 with SRST.

One thing you might want to try is a vpn tunnel between the 2811 and another router at your office/home location. The VPN tunnel will just pass thru the netgear. For example, you could setup the DMVPN hub at work/home, then setup the 2811 as a spoke.

Use Routing rules to achieve what you want. Most likely the Forwared rule. Use call viewer when you make the call. When the call comes in, look at call viewer and tailor your rule to match it. Then point it to the Call handler of your choice. Routing rules is the way to go, check it out.

The IP Phone will work, but if the Dell switch does not do CDP, no voice vlan for you. The IP phone will default and use the native vlan for both data and voice. I have a similar setup at home, the phones will fail to negotiate a voice vlan via CDP, then will use the default vlan for both data...

Don't forget the card-type command to specify if it is for a T1 or E1. Then then network-clock-participate command, then the network-clock-select.

The things that worked out for my design is to also create conferencinng resources on the SRST router with the extra DSPs. The mixing gets done locally and will not need to go over the WAN. The other benefits is Meet-me resources is now locally also. Conferencing will be hindered if you...

As for the switch type in CCM, I used 5E8 Custom and it works. Try 5E8 if it does not work. As for the MGCP bounce, ensure MGCP packets are getting proper QOS for the MPLS. If anything, it may be IOS related. Most of my MGCP problems has been IOS related. A quick upgrade fixes the weird MGCP...

I have had this problem many times before with different Callmanager Clusters I have ran into. Restarting the RIS Data Collector on the PUB will fix this issue. No reboot of the servers are necessary. A reboot is better since these are Windows server, and reboots are needed once in a while anyhow.

One thing I would try is to identify where the problem is truly. If you have a PRI and it is on a Cisco router, prove to yourself how the call gets sent out compared to the same area code that works. Using debug isdn q931 for PRIs or debug voice ccapi inout for T1 CAS, see how the call gets...

It sounds like you have a solution for the conflict and willing to introduce 8 as the site code for the Main Office. You may want to see how this will scale, as single digit site code is too restrictive. The possible solution is to: Have the remote site dial 846XX to get to main site. Create a...

I believe you are missing a key component to make this work. You will indeed want to create a VM Profile, but you must also put XXXX under the "Voice Mail Box Mask" field in the VM profile you just created. This will get rid of the * as it gets sent to Unity. Use the same Pilot as your regular...

there are different ways to handle the call once it is in Unity. You can use the routing rules and make a forwarded rule that matches the dialed-number, or an combination, and send it to Sign-in. Call viewer will help in figuring out the Forwarded rule. Another option is to create a Call...

It sounds like the Call Admission Control is stoping you as per the bandwidth and codec used from Locations and Regions. If you are using G711, you could change it to G729 and it will give you more bandwidth to play with.In turn, will allow more calls to go thru. If you introduce IP RTP Header...

Below is an example of blocking an incoming call on an H323 Gateway. voice translation-rule 100 rule 1 reject /^5551212$/ voice translation profile call_block translate calling 100 dial-peer voice 1 pots incoming-called . call-block translation-profile incoming call_block call-block...

If you are using an H323 gateway for the incoming call, you can block the call. It may not scale well if you start blocking lots of calls.