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Remote office cannot hear caller

David Coelho
Level 1
Level 1

Remote office. CallManager 8.6 at the main office and a 2901 router in the remote office H323 trunk between the offices.

Extension dialing works. Calls into the office via outside line works. Calls from that office to the outside world, however, the employee cannot hear the person they called but the outside person can hear the employee.

It is a site to site connection with all ports open.

I did a debug isdn q931 on the router and set up a virtual phone here in the HQ. When I call the phone I get the debug messages. When I use the virtual phone to dial out, no debug messages. I cannot replicate the issue remotely here though.

Anyone have any guidence where I can look for a solution on this?

Thanks again all. Very helpful lot!

7 Replies 7

David Coelho
Level 1
Level 1

I was able to replicate the problem. The remote employee called my cell and the call completed. They could not hear me though but I could hear them.

There were no q931 debug messages though. q921 below.

001734: Jul 30 16:40:11.427: ISDN Se0/0/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=15

001735: Jul 30 16:40:11.431: ISDN Se0/0/0:23 Q921: User RX <- RRf sapi=0 tei=0 nr=17

001736: Jul 30 16:40:11.431: ISDN Se0/0/0:23 Q921: User RX <- RRf sapi=0 tei=0 nr=17

001737: Jul 30 16:40:11.431: ISDN Se0/0/0:23 Q921: User RX <- RRf sapi=0 tei=0 nr=17

I am lost....butstill looking

From the remote office voice subnet, can you try pinging the interface that is bound to H323 interface on the 2901 router? If pings fail, you will have to fix that first.

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Pings look fine. Phones are registering properly so I do not think it is layer 3 but I am not completely certian.

The calls do connect, just half the audio is missing

Hi David,

When a phone registers to UCM, it needs to be able to connect to UCM. However, when a call is setup, only the signaling messages go to UCM, the media flows directly between the endpoints ie. in your case between the phone and the interface that is configured for H323. Media doesnt flow through UCM in your case.

For example, if your remote phone subnet is 1.1.1.1 and your h323 commands are configured under a Loopback0 interface with address 2.2.2.2, please try pinging 1.1.1.1 from the router but have it source it from the loopback address. ie. use the command "ping 1.1.1.1 source lo0".

Also, I believe since you said its a point to point connection, I am guessing you are referring site to site VPN. Please check if RTP inspection is disabled if you have a firewall in the mix. Even though that shouldnt matter since internal calls work fine but worth a try.

Also, once the call is up, you can look at call statistics and see if the received/sent packets are incrementing. If it is, then the phone is receiving packets but not playing out the media. The same statistics can be seen on the gateway using "show call active voice brief:

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Thank you for your time first off.

vrsksrst01#ping 10.13.163.24 source 10.13.254.163

Type escape sequence to abort.

Sending 5, 100-byte ICMP Echos to 10.13.163.24, timeout is 2 seconds:

Packet sent with a source address of 10.13.254.163

!!!!!

Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/4 ms

10.13.163.24 is the IP address of the phone located remotely and .163 is the H323 int on this router.

I made the call and did that show command but I get nothing. It is like the call is not being router through that router at all. No ISDN information either.

This is the show stats while the call was connected:

vrsksrst01#show call active voice brief

: ms. () + pid:

  dur hh:mm:ss tx:/ rx:/ dscp: media: audio tos:

IP : rtt:

  delay://ms

media inactive detected: media cntrl rcvd: timestamp:

long duration call detected: long duration call duration : timestamp:

  MODEMPASS buf:/ loss /

   last s dur:/s

FR [int dlci cid] vad: dtmf: seq:

  (payload size)

ATM [int vpi/vci cid] vad: dtmf: seq:

  (payload size)

Tele (callID) [channel_id] tx://ms noise: acom: i/o:/ dBm

  MODEMRELAY info:// xid:/ total://

         speeds(bps): local / remote /

Proxy :

bw: / codec:

  tx:

rx:

Telephony call-legs: 0

SIP call-legs: 0

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 0

This means that the call didnt go through the gateway that you are looking but a different gateway. You can go to Device -> Gateway and see if there are other gateways configured and go through each one of them and see if the call hits that gateway. You could also look at CSS/Partitions/Route Patterns/Route List/Route group to see which gateway the call goes through.

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Getting there now. Got it to go to the proper router now with your help. Getting busy signals but I think I can work this out now that I got it going right

THANKS

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