Issues with a SIP phone

Unanswered Question
Jun 16th, 2009


I am having a problem with SIP phones in my environment. The phones themselves are from Lifesize (part of a Lifesize room 200 and Lifesize team 200 units) setup as SIP phones. I am running UCM 6.1.3, a 3745 and a VG200 setup as h.323 gateways to the PSTN. All inbound and outbound calls hit the 3745 first and roll over to the vg200 if necessary. I am using the UCM publisher/master as the SIP proxy.

The phones work great for all internal calls (both to and from the devices and to and from all our standard Cisco endpoints). The video conferencing on the Lifesize units also works perfectly.

The issue arises when I attempt to make a standard voice call from the units to the PSTN or when attempting to call the assigned numbers from the PSTN. The outbound calls connect but only outbound audio works. Inbound calls never get presented to the phones.

I feel like I am missing something pretty simple but have been unable to figure out what.

Thanks in advance for any help.


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parshah Tue, 06/16/2009 - 10:29


Can you please explain a bit more in detail about this issue.

I am not sure what you mean by "The outbound calls connect but only outbound audio works. Inbound calls never get presented to the phones"

Do you mean when SIP phones call PSTN number, the call connects but there is one-way audio?


thompson.josh Tue, 06/16/2009 - 10:55

here are the two situations.

1. PSTN call from the SIP phone. I call a cell phone from the SIP phone, the cell phone starts to ring. The call is then answered from the cell and the SIP phone states that the call is connected. At this point the person on the cell phone can hear everything I say but I do not hear a word from their side. The call status app on the SIP phone shows packets being sent out but shows zero packets being received.

2. Call from PSTN to Sip phone. I have assigned the SIP phone a DID extension via call manager (both the SIP phone and UCM show registered). I have a person dial that number from a cell phone. The cell phone shows the call being placed and rings. The SIP phone does not ring. I have let the call ring for over a minute but the SIP phone does not show a call coming in. To makes sure it was not a problem that the PSTN was not presenting the call to my gateway; I assigned the DID to a standard Cisco phone, the call connected without a issue.

Hope this helps to clarify the issue.

parshah Tue, 06/16/2009 - 12:04


That helps a lot. So for scenario 1, one, have you check 'MTP required' on the SIP phone config page in CUCM? If not please check that. If its already checked please uncheck it.

For the second scenario, this could be an issue with CUCM or with SIP phone. To be best able to help you, can you please upload the detailed callmanager SDI traces for both these tests. Please be sure to enable sip traces in the trace config page first.


thompson.josh Wed, 06/17/2009 - 06:02


I went ahead and tested the "MTP required" settings. It was unchecked to begin with. After checking the box, I tested and the results were the same. As far as logs go, we use a SIP trunk to connect to our VM system. This results in some very cluttered SDI traces. It will take me some time to par them down but I will post them when i get a chance. I also opened a TAC case on the issue to see what they have to say. Will keep everyone updated.

Thanks for the help.

thompson.josh Mon, 06/22/2009 - 05:37


Here is what TAC and I have found out so far.

Step 1 -SIP phone sending 60056 as its listening port

Step 2 -MTP sending RTP packet to SIP phone port 60056

Step 3 - SIP phone sending ICMP port unreachable to MTP

Looks like it is going to be something with the SIP phone itself. Has anyone else had a issue like this?


parshah Mon, 06/22/2009 - 09:28


Can you check the following

1. SIP phone can ping the MTP ip address and vice versa?

2. There is no firewall/ACL between SIP phone and MTP.

I am assuming you are using the software MTP on the CUCM servers.


thompson.josh Tue, 06/23/2009 - 05:39

1. the phone and the mtp can ping each other

2. there is not a firewall/acl in between

3. we are using the standard (non enhanced) software MTP that is included with CUCM.

thanks for the help

parshah Tue, 06/23/2009 - 09:25


Can you get a packet capture from the switchport of the SIP phone for this issue?


thompson.josh Tue, 06/23/2009 - 11:51

Got it,

Thanks for all your help. I ended up turning off the SIP proxy and it is now working properly.



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