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One way audio

I am having one way audio issues with a CM 3.1.2 and Unity 3.02 install. I have tested network connectivity from unity to an IP phone and from a workstation attached to an IP phone to the unity server with success. I can call the DID directly and I have no problems. When I dial the main number and Unity opening greeting answers, I press "0" and Unity says "please hold while I transfer the call". There is dead air (no ringback) then the phone gives a fast busy (do not hear voice mail greeting)....if someone answers the IP phone, they can hear me, but I cannot hear them (DTMF digits from IP phone to me work, however). This is the case if I dial any other extension from the opening greeting menu. Usually this is a default gateway issue, but I have checked and everything is pointed to the same default gateway (3640 router). I am running 12.1.5T9 on the router. I am running dual CM's...I have dial-peer preference 0 going 192.168.2.3 (subscriber CM) and preference 1 going to 192.168.2.2 (publisher). I have verified the phones are active on the subscriber CM (192.168.2.3). My default gateway is the 3640 router at 192.168.2.1 (I can ping this from any device on the network (unity, cm, workstation). The default gateway for unity, CM 1, CM2, and the DHCP scope is also 192.168.2.1 (3640). I am using a 3524 switch with a single subnet. All ports are configured for dot1q trunking, portfast, dot1p tagging, extend COS 0, and trunk mode with the exception of the port connected to the router which is set for dot1p tagging, portfast and extend cos 0. <br><br>I have also verified the TSP configuration (primary CM is 192.168.2.3 (sub) and failover to 192.168.2.2 (pub), in addition to running the "test" to see connectivity between Unity and CM.<br><br>Any ideas?<br><br>

4 REPLIES
Gold

Re: One way audio

OK you have more than one issue here.

The first is no ring back on transfer. Check out these posts:

http://avforums.isomedia.com/cgi-bin/showthreaded.pl?Cat=&Board=unityent&Number=4510&page=&view=&sb=&vc=1#Post4510

You also have an issue with the operator extension forwarding back to Unity incorrectly. Make sure you have a valid CFNA and CFB destination on that line. Also make sure that the Calling Search Space looks in the Partition that the Unity ports are in if applicable.

Now for the one-way audio issues. Couple things you can do here:

1. If you have a 7960 or 7940 you can hit the 'i' button twice while you are getting one-way audio and check the RxCnt and TxCnt values. They should both be incrementing. If they are not then check the other phone so see if it thinks it is send and receiving packed. If you have a discrepancy then the packets are getting lost in the network. If not you might want to get a sniffer trace from the phone to see where it is sending the packets to.

2. Try placing the call on hold while you are getting one-way audio and then taking it off hold. Putting the call on hold tears down the RTP stream and forces it to be rebuilt. Does the audio return to two-way?

Keith

Keith Chambers
Unity Technical Lead
Unified Voice Team, San Jose
Cisco Systems
kechambe@cisco.com

Anonymous
N/A

Re: One way audio

I neglected to add that I can ring the IP phone when transferred from the "0" operator transfer during the opening greeting. The operator transfer function is working properly. All calls that are transferred from Unity are exhibiting this problem. Also, let me reiterate, even though I do not have audio from the IP Phone to the external caller, a DTMF tone (1 for example) can be heard when pressed from the IP phone.

The network is pretty simple....one 3640 router with one MFT-T1 port and a dual ethernet port, one 3524 catalyst switch with the configuration indicated in my original post. Typically one way audio issues are related to the default gateway...every device, including Unity, has a default gateway of 192.168.2.1 (ethernet port of router). The IP phones indicate this default gateway from the network configuration button on the phone. I can ping every device on the network from every location.

Since calls directly to Call Manager seem to behave properly (inbound DID), I am leaning towards a Unity issue. Unity is not supervising the transfers (release to switch is selected).

Any ideas?

Gold

Re: One way audio

Aaaaahhhh -- you left out this part 'even though I do not have audio from the IP Phone to the EXTERNAL caller'. I bet your don't get this problem on internal calls.

You need to add this command to the physical ethernet interface that you want to source all packets from on the router:

h323-gateway voip bind srcaddr **ip_address**

This command sets the source IP address to be used for this gateway. Right now you are doing a round robin back and forth between the ethernet interfaces.

Also, just so you understand what role Unity plays in this I will explain how a Unity transfer works. Unity is nothing more than a bunch of phones to the CallManager. So, when a call comes in to Unity and the caller instructs Unity to transfer, Unity hit the transfer button, just like a phone would, dials the extension and then hits transfer again. You will be able to reproduce this problem even if you take Unity out of the mix.

Keith

Keith Chambers
Unity Technical Lead
Unified Voice Team, San Jose
Cisco Systems
kechambe@cisco.com

Anonymous
N/A

Re: One way audio

That was it, thanks for your help.

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