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Unread 09-02-2007, 10:48 AM   #11
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It's not mis-routing. See my explanation here:

http://forum.voxalot.com/sip-broker-...html#post12316

.
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Unread 09-23-2007, 06:25 PM   #12
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Default nat option in asterisk configuration

Hi Martin, I hope you are still monitoring this thread. It has become somewhat cluttered. Misrouted ACK seems to be a common problem but it can have various causes.

Going back to my specific problem, I am hoping you can tell me if the asterisk PSTN gateways are using the NAT option in their configuration files. I have been doing some research on asterisk and I believe this option could have a bearing on my specific case.

The NAT options are as follows --
NAT undefined means default to NO
NO means use the RFC3581 extension ( use rport parameter )
ROUTE means do not use rport and work around bug in Uniden phones
YES means use rport and work around bug in Uniden phones
NEVER means do not use rport and do not work around bug in Uniden phones.

I realize things are very busy with the rollout of new features and a new business model.
Thanks in advance.
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Unread 09-23-2007, 11:39 PM   #13
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Hi telenerd,

The NAT setting is undefined in the configs of our PSTN gateway servers.

Quote:
Originally Posted by telenerd View Post
Hi Martin, I hope you are still monitoring this thread. It has become somewhat cluttered. Misrouted ACK seems to be a common problem but it can have various causes.

Going back to my specific problem, I am hoping you can tell me if the asterisk PSTN gateways are using the NAT option in their configuration files. I have been doing some research on asterisk and I believe this option could have a bearing on my specific case.

The NAT options are as follows --
NAT undefined means default to NO
NO means use the RFC3581 extension ( use rport parameter )
ROUTE means do not use rport and work around bug in Uniden phones
YES means use rport and work around bug in Uniden phones
NEVER means do not use rport and do not work around bug in Uniden phones.

I realize things are very busy with the rollout of new features and a new business model.
Thanks in advance.
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Unread 09-24-2007, 02:22 AM   #14
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Hi Martin
Unfortunately that is not the answer I wanted to see. My research led me to believe if the NAT option was YES or ROUTE then the CONTACT URI would be ignored.

So we still do not know why the ACK is not sent to my CONTACT URI when the call originates from the sipbroker PSTN gateway.

Most people will not have a problem if the ACK is sent to the URI that was registred because it is usually the same as the CONTACT obtained from the 200 OK. Not so in my case.

Is there no hope of getting this to work? I believe that RFC3261 is pretty clear that the ACK should be sent to the CONTACT URI derived from 200 OK.
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Unread 09-24-2007, 09:32 AM   #15
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Looking at your log snippets, my guess is the reason the ACK is *not* being sent directly to the Contact URI is because Voxalot NAT handling is kicking in and changing the Contact URI.

If the have "Symmetric NAT Handling" set to yes and your Contact URI is and RFC1918 address as highlighted in your log:

Contact: <sip:246875@192.168.1.3>

Then Voxalot will change this Contact in the 200 OK message before passing it back to the PSTN gateway.

Quote:
Originally Posted by telenerd View Post
Hi Martin
Unfortunately that is not the answer I wanted to see. My research led me to believe if the NAT option was YES or ROUTE then the CONTACT URI would be ignored.

So we still do not know why the ACK is not sent to my CONTACT URI when the call originates from the sipbroker PSTN gateway.

Most people will not have a problem if the ACK is sent to the URI that was registred because it is usually the same as the CONTACT obtained from the 200 OK. Not so in my case.

Is there no hope of getting this to work? I believe that RFC3261 is pretty clear that the ACK should be sent to the CONTACT URI derived from 200 OK.
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Unread 09-24-2007, 04:25 PM   #16
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Default

Quote:
Originally Posted by martin View Post
Looking at your log snippets, my guess is the reason the ACK is *not* being sent directly to the Contact URI is because Voxalot NAT handling is kicking in and changing the Contact URI.

If the have "Symmetric NAT Handling" set to yes and your Contact URI is and RFC1918 address as highlighted in your log:

Contact: <sip:246875@192.168.1.3>

Then Voxalot will change this Contact in the 200 OK message before passing it back to the PSTN gateway.
My voxalot Symmetric NAT handling is set to NO. The 600 echo test works fine.

As I mentioned in an earlier post, eventually an ACK arrives with the proper REQUEST URI. Up to that point there is two way audio. So with the arrival of a proper ACK the phone is satisfied but incoming audio ceases. Is it possible that voxalot is changing the media address to a private address as well? Private addresses can not be routed through the internet. My outgoing proxy is associated with an rtp proxy/media relay and the SDP contained in the 200 OK specifies a public address.
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Unread 10-25-2007, 05:09 AM   #17
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Hi Martin
After some weeks I decided to try once again to access my SIP phones through the PSTN gateway. Something apparently has been changed. After INVITE/200 OK the ACK now gets sent to the phone that answered and I have two-way audio.

There is one peculiar thing though. Immediately after the call is established the gateway sends a re-INVITE. It does not change anything.

Thank you. This will make Voxalot/Sipbroker much more useful for me.
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Unread 10-25-2007, 07:16 AM   #18
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Quote:
Originally Posted by telenerd View Post
Something apparently has been changed. After INVITE/200 OK the ACK now gets sent to the phone that answered and I have two-way audio.

There is one peculiar thing though. Immediately after the call is established the gateway sends a re-INVITE. It does not change anything.
Yes it did. See:

http://forum.voxalot.com/sip-broker-...html#post13367

The re-invite allows for a more direct audio path.
.
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Unread 10-25-2007, 04:43 PM   #19
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Quote:
Originally Posted by martin View Post
Yes it did. See:

http://forum.voxalot.com/sip-broker-...html#post13367

The re-invite allows for a more direct audio path.
.
I see. A more direct path at the gateway end.
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Unread 01-17-2008, 02:19 PM   #20
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Default SIP PSTN Access number not working.

Hi Telenerd,
Did you notice that, calls from Vancouver PSTN Access Numbers are not working again.? Calls drop with in few sec. And also that, PSTN access numbers from other destinations are also not working properly. All calls are dropping with in 2.5 mins. How do we solve this problem. Who is responsible to fix this issue?
I hope someone does it quickly.

Cheers.....
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