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Transfer between SIP Trunks

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Bonker1974

Bonker1974

Technical User
Oct 11, 2011
359
BS
Hi I have a BCM400 at the head office and a BCM 50 at the remote office connected via sip trunks.

The problem that i am having is every time a user on the 400 transfer a call to a user on the BCM 50 the call drop the calling party never get to speak to the user on the BCM50.

It,s not remote access packages because some calls transfer on and some calls just drop.

Allow Redirect is enable on all sets.
 
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Can you clarify? In your second statement you say every time a user transfers the call and the last statement you say some calls transfer and some don't.
 
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When a user transfer 10 call 3 out of 10 will transfer and the calling party connects to the user.

The other 7 calls might drop once the user press the join button on the 1120 IP phone. Or if the call is transfer you get no audio on the user side and the calling party side.

 
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I just realize the call is not actually dropping when it is transfer. What is happening is when the call is transfer there is no audio so it make is same like call is drop.
 
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Check your default gateways and make sure the correct ports are open on your firewall. Setting up a call uses different ports than the voice packets. Also, when this is happening, check to see if there is a duplicate ip address on your network.
 
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The default gateway is correct this is one network.

We have a local loop fiber connection between 2 sites. The connection form the cable company plug directly in to the switch at the remote site and into the switch at the head office no route are involved only the gateway to the internet.

The no audio on transfer only happen when a call come in to the head office and is transfer to the remote site.
 
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Do you have IP trunks to the PSTN as well and are the calls that are being transferred invole any kind of ip set? If so, you might want to look for codec mismatches and different packetization times
 
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