Voice over IP

Dear all,We have voIP between two sites, connected over MPLS and voice calls drop now and then. We use Avaya for Voice boxes and Cisco routers/switches elsewhere. QoS is configured on both ends of the MPLS to prioritize voice packets and I do not see...

Dears,As you know, RTP uses port range 16384 to 32767 dynamically. Can we define any specific static port manually so that rtp always use that udp port for voice call. I am using 3800 series router for voip setup. I tried to find any configuration bu...

Dears, I connected cisco 3800 router with ericsson MD110 exchange using QSIG over E1 Pri. Calls are forwarding properly over ip network but few advance features are not being enabled like auto callback. In existing scenario, User A-----MD110 PBX-----...

I'm running v.2.1 for the ip communicator. Every since I upgraded from 2.02 I have had a lot of complaints from users stating that they can't dial 800#. all other numbers work. If I un-check the "Optimize for Low Bandwidth" it works. user are able...

Hello all,We are currently running Call Manager, version 3.3(3) (Upgrading next month to 5.0)Is there any way to specify the caller id to be displayed for outgoing calls for specific extensions, instead of the corporate number now that it displays?Th...

Hi-We have an AT&T DSL on our remote office. When I tried connecting the 7960 directly behind the DSL modem (AT&T provided) my 7960 has either problems registering or getting kicked-out by the server after a few minutes.But when I installed a Linksys...

Hi,I've got two routers connected by ethernet to each other.Whereas I want to simulate a 1Mbps link between these two I have put a 1meg rate limit as input & output in connecting interfaces.rate-limit input 1000000 ....rate-limit output 1000000 ....T...

Hi, I setup one AS5400XM the incoming dialpeer and outgoing dialpeer are configured. we use the codec of outgoing dial peer is codec g729r8 bytes 80. but it cannot adjust bayte szie to 80. it still uses the default vaule (20) to send out traffic. any...

I have three PRI's running onto my router, with around 500 phone numbers on it. If there is a phone number on the PRI from teh telco but no associated DP on the router, then the call takes up all remaining trunks as it searches for a home.The fix fo...

I have a 1 Call Manager cluster with Unity and off box Exchange with 15 remote offices running off of this. Running G729 to the Unity Cluster, and all transcoding is done at the cluster site on CMM blades. My question is at what point is transcodin...

Hi,I use ipipgw ios to route the sip call to h323 gateway. The codec of inbound dial peer (sip) is different from the codec of outbound dial peer (h323). it does not work. the call gets fast tone. is it a bug of ipipgw ios?if the codec is same, the c...

i configued h.323 gateway (gateway is connected PSTN through FXO) behind internet NAT router and try to call that gateway from a softphone through internet. the dialed PSTN no is ringging but no voice for both ways. Pls refer the attached configurati...

Please help me with a software tool which is capable of recording .au files comply to cisco standards.

Im using 2811 with NM/HDV2-E1/T1 (PVDM2-32) for FOIP. Can anyone tell me what are the parameters should i use to calculate resource limitations for FOIP service.