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No Autio on Transfer sip trunk calls

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Bonker1974

Technical User
Oct 11, 2011
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Hi All
I have an issue with SIP trunk calls from Net to phone. Anytime you received a call on the SIP trunks it works find, the minute you transfer that call to a user you lost the audio.

Any help or suggestion would be great.
 
IS IP Office setup to send Re-Invites with updated SDP info? When a transfer happens an update or re-invite is sent which should encompass the RTP port currently used for the media stream.
 
I'd think that whatever IPO is doing to switch up that audio - however it does it - with a reinvite or whatever is definitely not jiving with what your provider wants/expects.

If you can post a monitor trace of the transfer and what the provider responded with, might give a clue to ask a more pointed question to your provider.

Also, turn off direct media path on 2 sets and try the xfer again - bet ya if media is pinned to IPO the whole time that you can transfer fine.

 
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