background noise, hiss on voip connection, 2811

winston123 · ‎04-28-2008

Guys, any ideas what it may be,i am getting background noise, hiss on the line, I use the vwic, PRI,router connects to tdm/pri line,voip calls are sent to public internet via 3560 switch.

as a troubleshooting measure, i anable Qos in the 2811, i am using voip only on this router, no data, no video.

*****************

class-map match-any voice-control

match ip dscp af31

match ip dscp cs3

match access-group 101

class-map match-any voice-RTP

match ip dscp ef

match access-group 100

!

policy-map Voice-QoS

class voice-RTP

priority 8kbps

set dscp ef

class voice-control

bandwidth 8kbps

set dscp af31

class class-default

fair-queue

!For RTP

access-list 100 permit udp any any range 16384 32767

access-list 100 permit udp any range 16384 32767 any

!For H.323

access-list 101 permit tcp any any eq 1720

access-list 101 permit tcp any eq 1720 any

a.cruea1980 · ‎04-29-2008

You say you're using a PRI, but your VoIP calls are sent to the internet via a 3560. . .

What calls are hissing? The calls using your PRI, or the ones going over the 'net?

Either way, there are a couple of things to check:

Does the hissing happen on all phones?

If you can hook up an analog phone via an FXS port, do so and try to make a call. Does that hiss?

You can try modifying the echo-cancel in your 2811 (this may eliminate some background noise).

You can also contact your telco and have them check the resistance on your PRI line and make sure your system is matched up to theirs. You can also change your gain on your 2811 so the echo-cancellers can do a slightly better job distinguishing echo (or background noise) from the voice signal. Be forewarned though, that this can causes the volume of a call to be loud.

If you could, at least answer the top questions I posed and we can probably help a little better. A lot more information is needed to better diagnose what could be happening.

winston123 · ‎04-29-2008

Sorry for the confusion,network layout

voice server---->router2811--->switch3560-->fwasa5510--->internet.voice server connects to router thru a PRI line,TDM calls are originated from the voice server thru a software app to the router...

question:where do I change the gain and echo on my 2811, what is the command string,and/or config mode level.

Thanks a lot

a.cruea1980 · ‎04-29-2008

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080149a1f.shtml

Check out that documentation. All the stuff I've listed to check they have there.

How exactly does your voice server connect to a PRI without a gateway? I've never seen that on CallManagers.

winston123 · ‎04-29-2008

Thanks man, this is great, i am still troubleshooting the echo,impedance, gain issue.

answer to your question:does your voice server connect to a PRI without a gateway, this is a back to back connection, I connect the vwic from the 2811 thru rj45 to a T1 dialogic card on the voice server, there is a software-based app that will originate the tdm call to the 2811 which will then send it out as voip thru fa0/1 which is connected to a 3560 switch for internet access.

Please advise, thx..

a.cruea1980 · ‎04-29-2008

I can safely say I've never seen that.

Generally, I've seen that your 2811 is the termination point for the VoIP part of the call, and the 2811 have a VWIC that hooks up to a PRI and puts the call out into the telco's network.

I realize this is a stupid question, but do you mean to tell me ALL your calls go out over the 3560? Are you working in a satellite office? How do your calls get onto the POTS? I just ask, because if you're a satellite office sending your calls out over a WAN link, the problem may well be something you can't control at all.

winston123 · ‎04-29-2008

Not exactly,i make my calls from the voice server, it gets to the router thru the pri connection and router sends it out as voip to our provider thru fa0/1..in dial-peer voice voip, we have a target ip where all the calls get sent to, it gets the the provider voip gateway and they do the final call termination/routing...so basically, there is not telco/pstn involve in the call-originating phase, just when the called party happens to be a landline..