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Unread 07-28-2007, 06:37 AM   #1
telenerd
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Default sipbroker pstn gateway sends ACK to proxy rather than the UA

When using a sipbroker access number or webcall to my voxalot account the call is not established correctly resulting in no voice. In my setup I use an outgoing proxy because my phones are behind a NAT and also I use multiple phones with the same username.

I did a message trace. The incoming INVITE had three RECORD-ROUTE with lr on, including the one from my proxy. The phone that answered sent out the 200 OK with the same RECORD-ROUTE as well as its CONTACT. The far end now sent ACK. This is where the problem lies. The REQUEST URI of the ACK was for the AOR/proxy not the CONTACT. This is in violation of RFC3261.

The PSTN gateway is an Asterisk. A search revealed that his problem had been reported to Digium ( developers of Asterisk ) quite some time ago and had been supposedly fixed. The messages from this Asterisk did not reveal its version. This long ago trouble report describes the problem thoroughly and provides references to the pertinent sections of the RFC. Here is a link to that trouble report - http://bugs.digium.com/view.php?id=2687

I would like the moderator to pass this report to those that that can correct the problem.
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Unread 07-28-2007, 11:05 AM   #2
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Quote:
Originally Posted by telenerd View Post
When using a sipbroker access number or webcall to my voxalot account the call is not established correctly resulting in no voice.
You mention webcall here. Is the problem you describe only applicable to the SIP Broker PSTN gateway or do you have a problem with webcall also?
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