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Unread 08-08-2007, 11:01 PM   #1
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Unhappy Calls from PSTN dropped in 30 seconds. No ACK message.

Since the last two weeks I have faced a problem when receiving calls form any PSTN access number. The call is answered normally but is dropped after 30 seconds. I am using SJPhone and before disconnecting the call the phone displays the message "Awaiting acknowledgement". After 30 seconds the call is disconnected and SJPhone shows the message "ACK Timeout". I have tried different PSTN access numbers and the problem is always the same. Calls from other SIP devices work perfectly.

Thanks in advance.

Andre Silveira
Voxalot Number: 910198
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Unread 08-09-2007, 06:32 PM   #2
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You did not provide many details of your setup. If you are using an outgoing proxy or some sort of a SIP gateway to get around NAT you may be having the same trouble I reported on the Sipbroker forum.

See "sipbroker PSTN gateway misroutes ACK"
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Unread 08-20-2007, 01:07 AM   #3
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Default Case details

Hi telenerd, thank you for your response. I read the message you mentioned and my problem is very similar, but not the same. I am in a home network, there is no proxy, there is no problem with NAT, and, the most important, everything worked well untill the the last month (July). Since that time the incoming calls from PSTN are dropped after 30 seconds and the problem is the lack of ACK from the other end to my UA (SJPhone). Attached is a file with the following sequence of messages: INVITE/SD, 100 Trying, 180 Ringing, several (11) 200 OK/SD and then a BYE message from SJPhone to the other end. I really appreciate with you or someone else can help me to solve this problem. Thanks!
Attached Files
File Type: txt PSTN Incoming Call-No ACK.txt (17.9 KB, 10 views)
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Unread 08-20-2007, 02:55 AM   #4
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I did see this Acknoledgement problem / timeout after 30 seconds with SJ phone some time ago.
I believe it was corrected by forwarding some ports to the computer IP address running the SJ phone or DMZing that IP address.

I am assuming that you are behind a router setup with private IP address.
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Unread 08-26-2007, 11:34 PM   #5
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Hi kurun, thanks for your tip. Now it is possible to receive calls from PSTN normally. The ACK problem was solved and the solution was to create a inbound port forwarding to UDP and TCP port 5060 to one IP address on the router´s firewall.
But sincerely, I consider this not a solution, but a workaround because now there is a static mapping to a specific IP address. Some weeks ago everything worked smoothly without this configuration and was possible to answer calls on any IP address/machine on the network, as soon it was registered to voxalot.

I would like to know if this procedure is normal and really necessary or it is possible to have a dynamic condition as was before.

Thanks
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Unread 08-27-2007, 02:04 AM   #6
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Default SJ-Phone setup

I am actually quite fond of SJ-phone because of it's multiple profile capability, and tried quite extensively to sort out this issue, even contacting SJ-Labs.

Are you using the latest version (1.65) or the earlier version of SJ-phone (1.60.289a, which I typically use) ?

I do not seem to have the problem any more actually, even though I have not fixed my computer IP or forwarded ports to it. The problem seems to have gone away when I fixed the IP address of the SIP hardware, and forwarded the appropriate ports to those addresses. Not sure how it affected the NAT performance.

For whatever it is worth, my set up is as follows :
1) Voxalot account with single registration
2) User Domain / Proxy Domain us.voxalot.com, SIP Port 5060
3) Register with Proxy, Proxy is strict outbound, Unregister Contact Address only all Checked
4) On the advanced screen, Accept redirection replies, Expose software version checked, all else unchecked
5) DTMF - RFC2833 (all else default values)
6) STUN - stun.voxalot.com.au:3478
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Unread 08-28-2007, 11:56 AM   #7
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There seems to be a bug in the SJPhone STUN logic. To fix the problem *without* port forwarding simply UNCHECK the use discovered addresses in SIP box on the STUN tab.
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