Click Here To Visit SIP Broker  

Go Back   Voxalot / SIP Broker Support Forums > Voxalot Forums > Voxalot Support

Voxalot Support Support for the Voxalot service.

 
 
Reply
Thread Tools Display Modes
Unread 08-08-2007, 11:01 PM   #1
910198
Member
 
Join Date: May 2007
Posts: 34
Thanks: 2
Thanked 1 Times in 1 Posts
910198 is on a distinguished road
Send a message via Yahoo to 910198 Send a message via Skype™ to 910198
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
910198 is offline   Reply With Quote
Unread 08-09-2007, 06:32 PM   #2
telenerd
Junior Member
 
Join Date: Jul 2007
Posts: 23
Thanks: 2
Thanked 1 Times in 1 Posts
telenerd is on a distinguished road
Default

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"
telenerd is offline   Reply With Quote
Unread 08-20-2007, 01:07 AM   #3
910198
Member
 
Join Date: May 2007
Posts: 34
Thanks: 2
Thanked 1 Times in 1 Posts
910198 is on a distinguished road
Send a message via Yahoo to 910198 Send a message via Skype™ to 910198
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)
910198 is offline   Reply With Quote
Unread 08-20-2007, 02:55 AM   #4
kurun
 
Join Date: Sep 2006
Location: Toronto, Canada
Posts: 568
Thanks: 70
Thanked 147 Times in 115 Posts
kurun is a jewel in the roughkurun is a jewel in the rough
Default

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.
kurun is offline   Reply With Quote
Unread 08-26-2007, 11:34 PM   #5
910198
Member
 
Join Date: May 2007
Posts: 34
Thanks: 2
Thanked 1 Times in 1 Posts
910198 is on a distinguished road
Send a message via Yahoo to 910198 Send a message via Skype™ to 910198
Default

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
910198 is offline   Reply With Quote
Unread 08-27-2007, 02:04 AM   #6
kurun
 
Join Date: Sep 2006
Location: Toronto, Canada
Posts: 568
Thanks: 70
Thanked 147 Times in 115 Posts
kurun is a jewel in the roughkurun is a jewel in the rough
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
kurun is offline   Reply With Quote
Unread 08-28-2007, 11:56 AM   #7
martin
 
Join Date: Feb 2006
Posts: 2,930
Thanks: 528
Thanked 646 Times in 340 Posts
martin is a jewel in the roughmartin is a jewel in the roughmartin is a jewel in the roughmartin is a jewel in the roughmartin is a jewel in the roughmartin is a jewel in the rough
Default

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.
.
__________________
Martin

Please post support questions on the forum. Do not send PMs unless requested.
martin is offline   Reply With Quote
Reply

Thread Tools
Display Modes

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Forum Jump

Similar Threads
Thread Thread Starter Forum Replies Last Post
Free DID Numbers, Free Voip Calls, & more Voip Info amroe Voxalot General 54 01-13-2014 09:11 AM
Codec order for SIP - PSTN calls occamsrazor Voxalot Support 0 03-08-2007 10:05 AM
Calls got dropped within 30 seconds sip2 Voxalot Support 1 02-27-2007 09:05 AM
Forward calls from PSTN to mobile via Pennytel - HELP BJReplay Voxalot Support 0 12-22-2006 12:04 AM
What am I doing wrong here? code- Voxalot Support 16 10-20-2006 05:12 PM


All times are GMT. The time now is 10:08 AM.


Powered by vBulletin® Version 3.7.2
Copyright ©2000 - 2024, Jelsoft Enterprises Ltd.