GV + ATA + Sipgate + Sipsorcery - Stopped working

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What steps will reproduce the problem?
1. Pick up phone
2. Dial number
3. silence, then fast busy signal

What is the expected output? What do you see instead?
Sorry.. didn't understand the question (I'm not THAT technical)

What version of the product are you using? On what operating system?
Got me on this one too.

Please provide any additional information below.

----------------

EasternPA, I just posted a question/problem on your famous "HOWTO: Free Calling 
with Google Voice" thread, and then later, I saw your post from JANUARY up top, 
asking people not to post there anymore, and come here instead.

How embarrassing that I did not read that first. So sorry!

Anyway... this is what I posted over there:

Hello to all,

This past March, THANKS to this thread, EasternPA's docs and the help of many 
people, I successfully made my GV work with my ATA box at home + Sipgate + 
Sipsorcery.

All has been working fine with minor downtimes, but today I am having an issue 
that just won't get resolved, and I cannot figure out what's wrong.

Dialing OUT from my home phone won't work - I get a dial tone, dial the 
number... then there is silence for a few seconds, and then a fast busy signal.

Neither initiating the call from GV's website works, I select it to ring my 
Sipgate number and nothing happens. It says "Calling you" but my phone won't 
ring.

Everything looks fine on all websites...
Sipgate shows my phone ONLINE in green.
Sipsorcery sees my ATA under SIP BINDINGS, and under SIP PROVIDERS everything 
is Registered.

So I have no idea what is wrong. I turned off my ATA box, waited a few minutes, 
turned it on again... still nothing.. calls made still go silent and then a 
busy tone.

Any suggestions will be really appreciated!

------------------

It is 10pm now... phone hasn't worked all day. I did not change absolutely 
anything on my accounts. It was really like working one day, and stopped 
working the next.

The ONLY thing I can think of, was installing the GV plugin on my computer 
browsers in order to use GV with Gmail, but that has nothing to do with my 
cordless phone connected to the ATA, right?

Original issue reported on code.google.com by Xandr...@gmail.com on 7 Oct 2010 at 2:10

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Have you tried rebooting your ATA and your router? If not, it's the time to do 
it, cycle power off and on. If this didn't help, go to Sipsorcery website, 
"Console" and click on "Connect". Try receiving a call (you can initiate 
callback from GV website), select the messages that appear on Console screen, 
press Ctrl-C to copy them and paste them here.

GV plug-in won't help because it just automates the process of logging in to GV 
website and initiating a call from there. If you can't do it manually then the 
plug-in won't be able to do it, either.

Original comment by mte...@gmail.com on 7 Oct 2010 at 3:08

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Thanks mtelis for responding! I had also, yesterday, powered down my modem, 
router and voip box (even though the first time, I had only powered the voip 
ata). And still nothing.

As for the GV plug-in... I know it has nothing to go with using the ATA, but I 
just mentioned it because it was the only thing different that I had done, the 
same day that my phone stopped working.

And regarding the CONSOLE, I did initiate a call using GV's website, selecting 
Sipgate as my ring number, and I GOT THE CALL. Yesterday, not even that was 
working (only when I would choose CELL as the ring number).

But even though one thing worked.... the main thing is still not working, which 
is trying to make outgoing calls from my home cordless phone attached to the 
voip box. I still get dial tone, dial the number, get silence for a few 
seconds, and then a fast busy signal.

Here's the info from the SS Console after making that successful call from GV's 
website: (I replaced my username with XXX)

Monitor 14:50:47:558: basetype=console, ipaddress=*, user=XXXXXXXXXXX, event=*, 
request=*, serveripaddress=*, server=*, regex=.*.
NATKeepAlive 14:50:49:058 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
NATKeepAlive 14:50:59:292 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
NATKeepAlive 14:51:09:573 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
DialPlan 14:51:13:589 sip1: No dialplan specified for incoming call to 
XXXXXXXXXXX@sipsorcery.com, registered bindings will be used.
DialPlan 14:51:13:604 sip1: Forwarding incoming call for 
XXXXXXXXXXX@sipsorcery.com to 1 bindings.
NewCall 14:51:13:604 sip1: Executing script dial plan for call to XXXXXXXXXXX.
DialPlan 14:51:13:620 sip1: Commencing Dial with: XXXXXXXXXXX@sipsorcery.com.
DialPlan 14:51:13:651 sip1: Call leg is for local domain looking up bindings 
for XXXXXXXXXXX@sipsorcery.com for call leg XXXXXXXXXXX@sipsorcery.com.
DialPlan 14:51:13:667 sip1: 1 found for XXXXXXXXXXX@sipsorcery.com.
DialPlan 14:51:13:667 sip1: ForkCall commencing call leg to 
sip:XXXXXXXXXXX@72.188.120.248:5061.
DialPlan 14:51:13:667 sip1: SIPClientUserAgent Call using alternate outbound 
proxy of udp:69.59.142.213:5060.
DialPlan 14:51:13:667 sip1: Switching to sip:XXXXXXXXXXX@72.188.120.248:5061 
via udp:69.59.142.213:5060.
DialPlan 14:51:13:667 sip1: SDP on UAC call had public IP not mangled, RTP 
socket 204.155.29.57:18564.
DialPlan 14:51:13:808 sip1: Information response 100 Trying for 
sip:XXXXXXXXXXX@72.188.120.248:5061.
DialPlan 14:51:13:901 sip1: Information response 180 Ringing for 
sip:XXXXXXXXXXX@72.188.120.248:5061.
DialPlan 14:51:13:901 sip1: UAS call progressing with Ringing.
NATKeepAlive 14:51:19:761 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
DialPlan 14:51:21:682 sip1: Response 200 OK for 
sip:XXXXXXXXXXX@72.188.120.248:5061.
DialPlan 14:51:21:682 sip1: SDP on UAC response had public IP not mangled, RTP 
socket 72.188.120.248:8000.
DialPlan 14:51:21:682 sip1: Cancelling all call legs for ForkCall app.
DialPlan 14:51:21:698 sip1: Answering client call with a response status of 200.
DialPlan 14:51:21:807 sip1: Dial command was successfully answered in 8.14s.
DialPlan 14:51:21:807 sip1: Dialplan cleanup for XXXXXXXXXXX.
DialPlan 14:51:21:823 sip1: Dial plan execution completed with normal clearing.
NATKeepAlive 14:51:29:932 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
DialPlan 14:51:36:854 sip1: Matching dialogue found for BYE to 
sip:69.59.142.213:5060 from udp:69.59.142.213:5060.
NATKeepAlive 14:51:40:120 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.

Original comment by Xandr...@gmail.com on 7 Oct 2010 at 3:01

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Well, the Console trace is for incoming call and it looks perfectly okay (which 
is logical, because you did receive incoming call). In order to solve your 
issue with outbound calls, I need to see the Console trace for such an outbound 
call. Again, go to the Console and try making an outbound call from your 
telephone connected to the ATA. If you don't see any trace, it means that 
something is wrong with the ATA or its settings.

Original comment by mte...@gmail.com on 7 Oct 2010 at 5:03

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Hi mtelis,

Nothing happens on the Console when I try to place a call.

Monitor 16:14:08:013: basetype=console, ipaddress=*, user=XXXXXX, event=*, 
request=*, serveripaddress=*, server=*, regex=.*.
NATKeepAlive 16:14:11:810 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
NATKeepAlive 16:14:22:029 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
NATKeepAlive 16:14:32:247 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
NATKeepAlive 16:14:42:466 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.
NATKeepAlive 16:14:52:637 sip1: Requesting NAT keep-alive from proxy socket 
udp:69.59.142.213:5060 to udp:72.188.120.248:5061.

I am so puzzled by this... How can something be working just fine for 7 months, 
and then from one day to another... it doesn't anymore, without ANY changes to 
anything. 

Like placing a successful call from the ATA phone before going to bed... then 
sleeping... and the next morning it doesn't dial out anymore. How is that 
possible?

The ATA (along with Router & Modem) is connected to Backup UPS Surge Protector. 
There hasn't been any rain or lighting here.... but then again, if it has been 
damaged, it wouldn't work AT ALL... and it still does, but only to RECEIVE 
calls, but not MAKE them.

How frustrating!

Original comment by Xandr...@gmail.com on 8 Oct 2010 at 4:22

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I think something's wrong with your ATA (or its settings). Could you install a 
proven softphone (such as X-Lite) on your PC, configure it to use your 
Sipsorcery account and try placing a call? If this works, then the problem is 
definitely in ATA, you need to check its settings. If everything is fine there, 
probably your device has become inoperative :-(

Mike

Original comment by mte...@gmail.com on 8 Oct 2010 at 6:47

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mtelis, could it be Sipsorcery?

Is there another service that I can use Sipgate with, to test if the problem is 
at Sipsorcery?

Original comment by Xandr...@gmail.com on 8 Oct 2010 at 6:51

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Ooops... I wrote the previous post BEFORE I read your last reply. Hold on... 
will try X-Lite now.

Original comment by Xandr...@gmail.com on 8 Oct 2010 at 6:53

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YESSSS, outbound call worked with X-Lite and I did see lots of activity on the 
CONSOLE when I placed that call.

Wow!! Now what? How will I find what's wrong in the config of my ATA if I 
haven't touched that config since March? I am not even sure I remember what I 
did back then. I'm a simple housewife with "limited" tech savviness, but I try 
hard to learn.

Maybe I need a new ATA? Any recommendations? My ATA became mine once I canceled 
VoIP service with broadvoice.com, which I had since 2004. I think they replaced 
my ATA box once in 2006.

It's a SIPURA SPA-2100.

Original comment by Xandr...@gmail.com on 8 Oct 2010 at 7:09

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I don't have any experience with SPA2100 but I'm familiar with SPA2102 which is 
a newer model replacing 2100.

The very first thing to check is if the ATA is registered to Sipsorcery. Go to 
Admin / Advanced mode, click on Voice / Info and check if you see Registration 
State: Registered  for the Line you use to dial out via Sipsorcery.

There are too many settings that could affect ATA's ability to dial out... I'm 
not sure I can help from remote.

Original comment by mte...@gmail.com on 8 Oct 2010 at 7:51

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There is a great tool I use for Sipura debugging, it's called syslog server 
(slogsrv.exe). It was available from official website but not any longer. There 
are the other sources, for example:

http://forum.voxilla.com/cisco-linksys-sipura-voip-support-forum/spa3102-bgopen-
dialplan-26337.html

Original comment by mte...@gmail.com on 8 Oct 2010 at 8:22

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Hi mtelis,

I kinda of disappeared, but I had family come visit for a few weeks, so I 
wasn't on the computer much.

In the meantime, it has been a royal pain to have to go to the computer every 
time I needed to make a phone call, as I am still with the same problem... my 
home ATA receives calls from people calling my GV... my ATA receives call-backs 
from the GV website (when I initiate the call via GV website), but I still 
cannot dial OUT using my home phone.

So... I did go into the SPA-2100 Config page, and yes, it does show Registered 
for Line 2 (I can't use Line 1 because it's blocked by the VoIP company I used 
to use), but I have been using Line 2 successfully since March!

Wow... what a mystery this is... Everything working fine one day... and the 
next, not.

Wondering if I just need to buy a new ATA and be done with this... after all, 
this ATA is at least 5 or 6 years old... Don't know if hardware for ATAs have 
changed much in the past years where it would really impact quality of calls 
and bugs like these.

What do you think? Seems like I cannot get anywhere with trying to figure out 
why suddenly I cannot dial OUT anymore.

Original comment by Xandr...@gmail.com on 21 Oct 2010 at 4:15

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It could be a problem with your dialplan (for the ATA), or quite a number of 
other problems. For example, most certainly your ATA has been configured for 
automatic provisioning (firmware updates, settings update). I wouldn't be 
surprised to learn that your provider (the one you acquired ATA from) 
provisioned all their units with new settings effectively overriding your 
settings.

So, I'd rather start with unit reset and try to configure it from the scratch. 
If this doesn't help, get a new ATA.

Original comment by mte...@gmail.com on 21 Oct 2010 at 6:09

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Hi mtelis,

I guess I give up on my ATA. So I'll just buy a new one.

2 Questions though....

1) EasternPA's Docs recommend the Linksys PAP2T-NA, but I have heard of the 
SPA3102 also. Which one is best in your opinion?

2) What config do you use? As in... Modem - ATA - Router   or   Modem - Router 
- ATA??

For the longest time, I have been using Modem - Router - ATA, but today I got 
fed-up with dealing with a delay on my International calls, plus loss of Audio 
on one side (people suddenly can't hear me, or the other way around... 
sometimes total loss of Audio and then 10 secs later, it would come back. I 
always had to advise people on the beginning of a call, that if the call went 
silent, for them to just wait about 15 secs). I read somewhere that it had to 
do with Router packet loss.

So today I put the ATA in front of the Router... Modem - ATA - Router, called 
my grandfather in Brazil, and the call seemed much better. But... now I have 
another problem... Because the ATA is feeding the internet to my Router.. my 
computer net speed got slower. Speedtest.net showed 7mbps where before it used 
to be around 18-20mbps.

Aaaargh!!! Maybe the SPA-2100 ATA is just OLD.. and a new one won't slow the 
connection down??

So which config do you use? BTW, my router is Linksys G - WRT54GS

Thanks so much my dear... I'll be very happy once everything is working well 
again!

Alexandra

Original comment by Xandr...@gmail.com on 24 Oct 2010 at 7:11

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Hi Alexandra,

SPA3102 and SPA2102 (which is my preference over PAP2T) have different 
functionality. The first has 1 FXS and 1 FXO port and generally, you install it 
in between your telephone and telephone line outlet. SPA2102 has 2 FXS ports; 
you connect two telephones to it and you can't use it to manipulate your phone 
line, if any.

I don't have a separate ADSL modem, it's built-in my WiFi router. If you put 
SPA2102 in between the modem and the router, all your computers will be in 
double-NAT situation which may cause you some issues. So, I'd recommend to put 
ATA behind the router.

On the other hand, if you're getting better voice and double-NAT isn't giving 
you any trouble -- why not?! ;-)

Mike

Original comment by mte...@gmail.com on 24 Oct 2010 at 8:53

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Hi Mike,

Living and learning. Had to Google "double-NAT" and "FXS/FXO" to try to 
understand your post. Lol.

Anyway, I do not use the wall phone jacks in my house at all. Should I? I have 
a base cordless set/system with 3 headsets, so the base is the only one that 
has the phone cable that goes into the ATA.

Also, my internet connection is Cable.

So... I found this post on another website, where the guy says:

> I discovered recently that my VoIP router (SPA-3102) and its
> predecessor (SPA-2100) can't route from WAN-LAN faster than about
> 7.5Mbps. 

http://www.velocityreviews.com/forums/t594987-re-low-router-throughputs-wan-lan.
html

Hmmm... interesting eh?

Original comment by Xandr...@gmail.com on 25 Oct 2010 at 3:43

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If you don't have regular (PSTN) phone line you'd better go with SPA-2102. And 
yes, SPA's built-in router leaves a lot to be desired.

Original comment by mte...@gmail.com on 25 Oct 2010 at 5:04

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Mike,

Thank you immensely for your time and all your responses. I will end this topic 
now, though I have further questions and still some doubts regarding which ATA 
to buy. But thanks to your link for the Voxilla Forums, I will now post my 
questions there and relieve you from duty... LOL!!!

Thank you SO SO MUCH once again!!

Alex

Original comment by Xandr...@gmail.com on 26 Oct 2010 at 3:26

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You're very welcome!

Original comment by mte...@gmail.com on 26 Oct 2010 at 7:58

• Changed state: Done
• Added labels: Type-Other
• Removed labels: Type-Defect

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Hi mtelis Mike!!!!!!!!

I just came here because I was thinking of you!!! (you were so kind and 
responded to all my posts, so I didn't forget you)

Anyway... I also wanted to let you know what happened to the little Saga we 
discussed here.

I got to say that I couldn't be HAPPIER!!!!

I'll tell you why...

On this thread, at one point, you gave me the link to the Voxilla Forums.

Even after our conversation here, I was still on the fence about which other 
ATA to buy, and frustrated about that one-way-audio problem I had with my GV 
International calls.

So I went to the Voxilla Forums and posted a topic there.

Then I got frustrated that no one responded days later, so I just went ahead 
and ordered the SPA-3102.

I then expressed my sadness that no one responded, and wrote that I had ordered 
the SPA.

One user named "FALA", finally responded and told me to return the SPA, and 
purchase an Obi110!!!

I returned the SPA as soon as it arrived, but took me forever to purchase the 
Obi110 because it's always out-of-stock on Amazon.com. As soon as it's 
available, people run for it, and it goes out-of-stock again! (at least the one 
with the $49.99 price, because sometimes you find it for $80-$90 which means 
that people are trying to profit for it being out-of-stock).

Anyway... I received my Obi110 and it's AMAZING!!!!!!!!!! Awesome awesome 
awesome.

The setup of the Obi110 with Google Voice is a breeze!!! It took me DAYS to 
setup my SPA-2100 + Sipgate + Sipsorcery to work with Google Voice!!! And that 
was following EasternPA's Docs and getting a ton of help from other users on 
the Google Voice Help Forum.

So I had to come here and tell you about this Obi110 that is even cheaper that 
the SPA's.

Maybe you already know about it... but I leave my 2 cents here and thanks once 
again for everything.

PS: I don't know yet if I still have the Audio problem because I haven't made 
an International call yet, but I hope it is gone!!

Original comment by Xandr...@gmail.com on 15 Mar 2011 at 11:35

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I heard about Obi110. Judging by the specs, it's amazing device, indeed. 
Unfortunately, I don't have this device on hand and can't comment further.

Original comment by mte...@gmail.com on 17 Mar 2011 at 10:49

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That's ok! Didn't come here to ask you for comments, but to tell you about it 
in case you didn't know. 'Twas just because I thought of you since you helped 
me so much. That's all!! :o)

Original comment by Xandr...@gmail.com on 18 Mar 2011 at 7:26