5610 SIP Dect phones...ONE-way audio

[cpegeno]

I am currently working on an upgrade to one of my clients. They have 1 (one) Mitel5000 6.0.11.98, and are using multiple IP phones over 4 separate static IP networks on their site. I have all of the IP phones 5320 and 5330's NAT'ing to the system. All of them work fine. I installed 3 5610 SIP dect cordlesses at one of the locations. I get the "full moon" and such. BUT, during 21 separate test calls only 1 of them had two way audio (call #18).

My first question is: Does SIP require different ports opened to pass audio, other than the ports required for normal IP phones? If so what are they?

My second question is: With DMZ set at the firewall where the system resides, the audio is still one way, unless we change the ip scheme at the sip dect location. (i.e. the system ip location scheme is 192.168.1.XXX and so is the location of the SIP dect; to make the SIP dect phones work, we have to change the location where they sit to 192.168.2.XXX). Is there any other work around for this?

Please advise?
Thanks in advance!!

[cpegeno]

Everywhere i read port 5060. Is there a set or series of ports or is this the only one?

The Controller is on Network A. The sip dect phones are on Network B. Network A has been setup to open the ports on the firewall to allow for the IP phone traffic to pass. Network B is behind a Fios router (which is allowing the normal IP sets to communicate). Do I need to set up a distinct Port Forward rule for Network B in the fios router?

[Tech Electronics]

cpegeno,

SIP only requires port 5060 to communicate unless that has been changed in the SIP Phone Group Configuration.

I am not sure that I totally understand you second question, but I will provide an answer I think you are asking about.

The 5000 and the SIP phones are on the 192.168.1.xxx network and they work which means the phones are setup to work on the native network just fine. When you change the IP address of the SIP Phones to 192.168.2.xxx they no longer work, but did you change them from Native to NAT in their SIP Phone Group Configuration or are you allowing port 5060 to get through the firewall/gateway without it being "transformed"; sometimes this is a problem if the equipment is making changes to the packets being sent to or from the system.

The FIOS router would need to allow for SIP traffic on port 5060 to go through without it transforming the packets.

Thanks,

TE