One way audio

danielgoq · ‎04-07-2009

Hello everyone,

This morning we started having some troubles with the voice system. We are experiencing one way audio issues.

We have the following:

CUCM6.1 --- 3560SW -- 3825(8FXO) --- PSTN

|

VG224

Any IP Phone connected to the 3560 switch can make calls to anywhere (internal IP or Analog extensions or PSTN), however, even if the call is connected, the person on the other side can't hear me no matter if it is another IP Phone, an analog phone connected to the VG224 or on the PSTN.

On the other hand, if I start the call on an analog phone, everything works perfectly, I can call an IP Phone and they will hear me and I will hear them. Also, if I call anyone outside on the PSTN I will hear them and they will hear me.

No changes have been made to the configurations, everything was working just fine!

Using the analyzer tool, if I enter the analog extension as caller number and an IP Phone extension as called number, I'll get a route pattern result.

If I enter an IP Phone extension as caller number and an analog extension as called number, I'll get a block pattern result.

Both, the IP and the analog extensions are within the same range 23XXX.

If I enter the IP phone extension as caller number and a PSTN number as called number, I'll also get a block pattern result.

Does anyone have an idea of what could be happening??

Thanks a lot!

Nicholas Matthews · ‎04-07-2009

Your best bet:

Check 'stream statistics' on the phone webpage during the problem. Note the IP address it is sending to. Make sure that the default gateway of the phone can reach that address.

On the gateway use 'show call active voice brief'. You should be able to find the IP address of the IP phone in there. You can use 'show voip rtp connections' to make sure the gateway is showing the right source address when talking to the IP phone. If not, you would need an appropriate bind command:

sip:

voice service voip

sip

bind [media] [control] [both] ....

h323:

interface fa0/0

h323-gateway voip src-ip ....

mgcp:

mgcp bind [media] [signal] interface ....

hth,

nick

a.gooding · ‎04-07-2009

and you are sure nothing has changed on the switch from a routing standpoint?

routing, which is most of the times what causes this.

gateways are correct on the ip phones right? check the IP phone gateways

check back your routing (it wouldnt hurt) and see if there isnt anything there.

as metioned, check the stats as well on the phone. call one ip phone to another connected on the same switch (forget the pstn and vg for now since we know the VG is working.). when the call starts hit the help button twice and see if you have any packets passing. ill take a guess at saying you wont see anything.

im sure others would be helpful as well if it turns out not to be routing.

hope it helps a little bit at least.

just an FYI, i had a customer with a firewall centric network that caused one way audio when thier rules changed one evening. the voice engineers said nothing changed and from their standpoint nothing did......just to jogg the brain a little :)

danielgoq · ‎04-08-2009

Yes... Nothing has changed...

I rebooted the server (because everything seemed to be normal) and now we're able to connect call correctly but... conference bridges are failing:

SigmaTVoiceGW#sh sccp all

SCCP Admin State: UP

Gateway IP Address: 10.143.153.3, Port Number: 2000

IP Precedence: 5

User Masked Codec list: None

Call Manager: 10.143.153.4, Port Number: 2000

Priority: N/A, Version: 3.1, Identifier: 7825

Transcoding Oper State: ACTIVE_IN_PROGRESS - Cause Code: CCM_REGISTER_FAILED

Active Call Manager: 10.143.153.4, Port Number: 2000

TCP Link Status: NOT_CONNECTED, Profile Identifier: 2

Reported Max Streams: 10, Reported Max OOS Streams: 0

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30

Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code: CCM_REGISTER_FAILED

Active Call Manager: 10.143.153.4, Port Number: 2000

TCP Link Status: NOT_CONNECTED, Profile Identifier: 1

Reported Max Streams: 24, Reported Max OOS Streams: 0

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30

I don't understand what's happening

danielgoq · ‎04-08-2009

This is what the debug sccp is showing.

donovan222 · ‎04-10-2009

Transcoding Oper State: ACTIVE_IN_PROGRESS - Cause Code: CCM_REGISTER_FAILED

You need to go in to you DSP farm and do a (no SCCP), SCCP your DSP's are not registering with CCM. If you look under the Media resources and media termination points you should see your gateway unregistered.

DONOVAN RAAUM
SENIOR UNIFIED COMM ENGINEER
O 763-971-2433
E Draaum@convergeone.com