Puzzling SIP to H323 issue..

[MattRutter]

I have an IP500 at 9.1... IP phones and VM Pro server are on LAN 1 - all in the same subnet.. There is a SIP trunk on LAN2..

When I dial in on the SIP to voicemail, I can hear the prompts, and record and listen to greetings.. with no problems at all...

When I dial in to one of the IP phones, I get one way speech.. I can hear them, but they can't hear me..

The phones get thier addresses from the IP Office and the default gateway is set to the IPO itself...

How are voice packets failing to get out from the phones, but work from VM??

 

[JayNEC]

Lots of details missing in regards to subnets and addresses, but it's probably a routing issue. The first thing to try is turn off "allow direct media path" on a given test extension.
This will force the phone to hit the IP Office first rather than allow the call setup to tell it to "shuffle" to the trunks.

It's off by default on the SIP trunks themselves.

New England Communications

http://www.necomm.com

 

[StubbedmytoeonIPO]

DMP

 

[MattRutter]

DMP is off for my test phone....

 

[holdmusic34]

1) codec mismatch between sip and h323.
Nail it down to one codec only like g729 ulaw.
Check in ss or monitor what the trunks use when on a vm call and lock that down on a h323 call.

2) try a static ip address against the extension.

3) check what ports are being used for the sip call in the rtp stream.


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