PSTN disconnect after call is answered

wishfull · 09-28-2007, 04:57 PM

It was all working for last 2 years until about a week ago. Now when one call me using les.net pstn gateway call disconnects as soon as the call is answered. SIP to SIP works fine. Calls from some of other gateways I've tested work fine.
I don't know what to do anymore I have checked and double checked all the settings, hooked up the ATA (Sipura 3000) in a DMZ, even tried hooking it directly to the internet with any firewalls still no luck. Sent an email to les.net and sipphone.com no response either one. I have tried 2 other ATAs and same results. Please help I'm at a loss.

emoci · 09-28-2007, 05:58 PM

Which PSTN number?

It could very well be a problem with that PSTN, if it works from everywhere else...

wishfull · 09-28-2007, 06:38 PM

Quote:

Originally Posted by emoci

Which PSTN number?

It could very well be a problem with that PSTN, if it works from everywhere else...

NY 646-291-2168 (Les.net) doesn't work but 347-427-9019 1+# does work. Phoenix, AZ 602-427-5727 (Callcentric) is iffy sometimes it works not others.
I got a reply from les.net and they said it's SIP Broker PSTN gateway issue, I should be asking for help here. Is there someone who could look into this? I have failed log file, if that will help.

emoci · 09-28-2007, 07:01 PM

Well, I don't know if this helps, but I tried both numbers (646 numbers), and they do work for me....

Some things to look into:

-Before trying any of these other things, if you can, try some other Access Numbers and see if the problem persists with them too.

-The Symmetric NAT setting on your VoXalot account is it set to YES or NO
(if it's YES, leave it alone, if it is NO, try setting to yes)

-One problem that has been coming lately, is the support for ReInvites (the things is most Linksys Devices I've seen have it enabled by default)

-Also if you don't already, have STUN set up

-Also check your router, if it has UPnP enable it (it maybe that it's closing the port after the connection is established dropping the line) (If you have UPnP, I would turn off DMZ)

wishfull · 09-28-2007, 07:59 PM

ReInvites is set to 30. I have no idea where to turn it off. Also remember I did try hooking up the ATA outside the firewall and same results. I also have tired other SIPPHONE.COM numbers again same results. I'm starting to wonder if the problem is with SIP Broker and sipphone.com (1747) numbers. UPnP is enabled and DMZ is off.

emoci · 09-28-2007, 08:10 PM

Ok..., so you have an ATA registered with Gizmo then, I assume!!!!

Try getting a free VoXBasic account and registering that in your ATA, and see if you have the same issue ? (You may be able to isolate if it's a Gizmo/SipPhone issue or not)

Your ReInvite setting is the same as mine...(so that should be fine)

BTW, you tried just powercycling the ATA right (turn off, wait a bit, plug back in)

09-28-2007, 04:57 PM	#1
wishfull Junior Member Join Date: Sep 2007 Posts: 16 Thanks: 1 Thanked 0 Times in 0 Posts	PSTN disconnect after call is answered It was all working for last 2 years until about a week ago. Now when one call me using les.net pstn gateway call disconnects as soon as the call is answered. SIP to SIP works fine. Calls from some of other gateways I've tested work fine. I don't know what to do anymore I have checked and double checked all the settings, hooked up the ATA (Sipura 3000) in a DMZ, even tried hooking it directly to the internet with any firewalls still no luck. Sent an email to les.net and sipphone.com no response either one. I have tried 2 other ATAs and same results. Please help I'm at a loss.

09-28-2007, 07:59 PM	#5
wishfull Junior Member Join Date: Sep 2007 Posts: 16 Thanks: 1 Thanked 0 Times in 0 Posts	ReInvites is set to 30. I have no idea where to turn it off. Also remember I did try hooking up the ATA outside the firewall and same results. I also have tired other SIPPHONE.COM numbers again same results. I'm starting to wonder if the problem is with SIP Broker and sipphone.com (1747) numbers. UPnP is enabled and DMZ is off. Last edited by wishfull; 09-28-2007 at 08:07 PM.

09-28-2007, 08:10 PM	#6
emoci Join Date: Jul 2007 Location: Toronto, Canada Posts: 1,422 Thanks: 123 Thanked 369 Times in 282 Posts	Ok..., so you have an ATA registered with Gizmo then, I assume!!!! Try getting a free VoXBasic account and registering that in your ATA, and see if you have the same issue ? (You may be able to isolate if it's a Gizmo/SipPhone issue or not) Your ReInvite setting is the same as mine...(so that should be fine) BTW, you tried just powercycling the ATA right (turn off, wait a bit, plug back in) Last edited by emoci; 09-28-2007 at 08:17 PM.

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09-28-2007, 05:58 PM	#2
emoci Join Date: Jul 2007 Location: Toronto, Canada Posts: 1,422 Thanks: 123 Thanked 369 Times in 282 Posts	Which PSTN number? It could very well be a problem with that PSTN, if it works from everywhere else...

09-28-2007, 07:01 PM	#4
emoci Join Date: Jul 2007 Location: Toronto, Canada Posts: 1,422 Thanks: 123 Thanked 369 Times in 282 Posts	Well, I don't know if this helps, but I tried both numbers (646 numbers), and they do work for me.... Some things to look into: -Before trying any of these other things, if you can, try some other Access Numbers and see if the problem persists with them too. -The Symmetric NAT setting on your VoXalot account is it set to YES or NO (if it's YES, leave it alone, if it is NO, try setting to yes) -One problem that has been coming lately, is the support for ReInvites (the things is most Linksys Devices I've seen have it enabled by default) -Also if you don't already, have STUN set up -Also check your router, if it has UPnP enable it (it maybe that it's closing the port after the connection is established dropping the line) (If you have UPnP, I would turn off DMZ)