Sipbroker pSTN gateway misroutes ACK

telenerd · 07-30-2007, 11:01 PM

When using a sipbroker access number or webcall to my voxalot account the call is not established correctly resulting in no voice. In my set
up 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 two RECORD-ROUTE with lr on, including the one from my proxy. The phone that answered sent ou
t 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 a
go and had been supposedly fixed. The messages from this Asterisk did not reveal its version. This long ago trouble report describes the pro
blem thoroughly and provides references to the pertinent sections of the RFC. Here is a link to that trouble report -
bugs.digium.com/view.php?id=2687]0002687: ACK sent to wrong address - Digium Issue Tracker[/url]

I would like the moderator to pass this report to those that that can correct the problem.
Thank you

martin · 07-31-2007, 01:25 AM

When you say you are using the PSTN gateway which number(s) specifically are you referring to?

Some of you numbers are hosted by the sponsors themselves while others are hosted on our PSTN gateway server which is running Asterisk 1.2.1

You also mention webcall which is completely different and I would like to better understand this issue.
.

telenerd · 07-31-2007, 01:46 AM

I suppose I should have said virtual toll free. The PSTN access number I am using is 604-628-4266 which is in the Vancouver area of British Columbia. Incidentally the BYE message suffers the same fate as the ACK if the release is initiated from the PSTN end. The UA never receives it.

martin · 07-31-2007, 05:06 AM

Quote:

Originally Posted by telenerd

I am using is 604-628-4266

This is a LES.NET number that is hosted on one of our servers. As I mentioned, our PSTN access numbers are hosted on an Asterisk 1.2.1 installation.

Looking at the links you provided unless i'm mistaken the bug you mention should have been fixed in this version.

Am I correct in understanding that the ACK and BYE are hitting your proxy rather than contact host?
.

telenerd · 07-31-2007, 05:44 PM

You are correct. If I understand the RFC correctly when strict routing is in effect the contact URI would be the last URI in the route set and if loose routing is in effect the contact URI would be in the request line. In this instance the contact URI is missing entirely.

If I did this correctly there should be an attachment with this reply showing the message sequence as seen at the outgoing proxy.

telenerd · 07-31-2007, 05:52 PM

Let's try it with a file name extension of txt

07-30-2007, 11:01 PM	#1
telenerd Junior Member Join Date: Jul 2007 Posts: 23 Thanks: 2 Thanked 1 Times in 1 Posts	Sipbroker pSTN gateway misroutes ACK When using a sipbroker access number or webcall to my voxalot account the call is not established correctly resulting in no voice. In my set up 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 two RECORD-ROUTE with lr on, including the one from my proxy. The phone that answered sent ou t 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 a go and had been supposedly fixed. The messages from this Asterisk did not reveal its version. This long ago trouble report describes the pro blem thoroughly and provides references to the pertinent sections of the RFC. Here is a link to that trouble report - bugs.digium.com/view.php?id=2687]0002687: ACK sent to wrong address - Digium Issue Tracker[/url] I would like the moderator to pass this report to those that that can correct the problem. Thank you

07-31-2007, 01:25 AM	#2
martin Join Date: Feb 2006 Posts: 2,930 Thanks: 528 Thanked 646 Times in 340 Posts	When you say you are using the PSTN gateway which number(s) specifically are you referring to? Some of you numbers are hosted by the sponsors themselves while others are hosted on our PSTN gateway server which is running Asterisk 1.2.1 You also mention webcall which is completely different and I would like to better understand this issue. . __________________ Martin Please post support questions on the forum. Do not send PMs unless requested.

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07-31-2007, 01:46 AM	#3
telenerd Junior Member Join Date: Jul 2007 Posts: 23 Thanks: 2 Thanked 1 Times in 1 Posts	I suppose I should have said virtual toll free. The PSTN access number I am using is 604-628-4266 which is in the Vancouver area of British Columbia. Incidentally the BYE message suffers the same fate as the ACK if the release is initiated from the PSTN end. The UA never receives it.

07-31-2007, 05:44 PM	#5
telenerd Junior Member Join Date: Jul 2007 Posts: 23 Thanks: 2 Thanked 1 Times in 1 Posts	You are correct. If I understand the RFC correctly when strict routing is in effect the contact URI would be the last URI in the route set and if loose routing is in effect the contact URI would be in the request line. In this instance the contact URI is missing entirely. If I did this correctly there should be an attachment with this reply showing the message sequence as seen at the outgoing proxy.