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.
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.
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.
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.
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