Google Voice + SipGate + SipSorcery = VOIP Phone.. VERY UNSTABLE?

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What steps will reproduce the problem?
1. Dial w/ voice.google.com to SipGate phone
2. Phone may ring and it may not
3. Dial out w/ sipgate phone.  Rings with "fake tone" for awhile and sends
to busy tone.

Here's the issue.  I have EVERYTHING set up correctly (as far as I know). 
I have a modified RTP300 so that its not longer locked to Vonage.  I set
the preferences to exactly the Sipsorcery settings on the google document.
 In sipsorcery, its even registered.  As for the dial plan, I've used the
simplest one. 

However, it is EXTREMELY unstable.  I SOMETIMES get the incoming calls and
most of the time, I can't press "1" to accept.  I can have Google Voice
call the number and it will ring and work.  However, I cannot call out as
it "rings" for like 30 seconds and then goes busy.  It never "switches"
over to the actual google voice ring.  Is there ANY solution to this?  What
am I doing wrong here?  I also have gizmo5, but I always though Sipgate was
more stable...

Original issue reported on code.google.com by UCIR...@gmail.com on 19 Jan 2010 at 9:17

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Thank you for opening a ticket.

First, I haven't been able to update the document yet on exactly how to 
configure SIP
Sorcery and sipgate. Therefore, if you've followed the document to a "T", then 
you
will have issues. We found what appears to be a new resolution just a few hours 
ago
and I haven't sat down to fix the docs yet. Could you confirm that the Simple 
Dial
Plan you're using is the current one?

Finally, if you have kept up with the recent posts, could you please confirm 
for me
that you have 2 VoIP phones configured on just one sipgate account and that each
sipgate phone has its own line in the SIP Providers list as well as SIP Provider
Bindings? I don't know if it matters how they are named, but I did name them 
sipgate
line 1 and sipgate line 2. I also had multiple calls not ring today, so there is
something still going on, although outbound dialing is about 95% reliable for 
me.

Original comment by easter...@gmail.com on 19 Jan 2010 at 10:08

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Hi, yes I do have the simple dial plan.

However, I do not have 2 VoIP phones configured on the one sipgate account - 
just
one.  I didn't think it was necessary to have two of them?

What is the best provider in doing this Google Voice --> VOIP?  I also have 
Gizmo5,
but that's been known to be unstable as well.  I can use any provider and will 
set it
up, I just want a stable environment for outbound and inbound.

Thanks for the help!

Original comment by UCIR...@gmail.com on 20 Jan 2010 at 1:06

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Hey thanks for checking in. Unfortunately, I don't have any setups to offer 
right now
that are as stable as what we've experienced historically. SIP Sorcery 
underwent some
pretty significant changes recently, and no one seems to be able to nail down 
the
smooth calling we've had for a while. In the meantime, please read this message 
I
posted earlier today. It describes exactly how to modify your sipgate and SIP 
Sorcery
accounts to account for the changes that recently occurred to the SIP Sorcery 
servers. --

Log into your sipgate account. Click Settings in the upper right. Click "+ VoIP
Phone" at the top of the right menu. Confirm adding the phone. You will be 
shown a
new set of SIP credentials for this second phone. Take those credentials down. 
Return
to SIP Sorcery. Go to the SIP Providers page. Modify your current sipgate 
account to
show "sipgate line 1" and update the Register Contact field by replacing
"@sipsorcery.com" with "@174.129.236.7".

Now add a new SIP Provider, named "sipgate line 2". Enter the new SIP 
credentials
sipgate just showed you for your new VoIP phone, activate the Register 
checkbox, and
duplicate the Register Contact from line 1 except ending in "@174.129.234.254".

This will make any call coming in to your sipgate account ring on *both* SIP 
lines
simultaneously. Since SIP Sorcery moved to a 2-server setup, the return call 
coming
from Google Voice through sipgate can land on either SIP Sorcery server, even 
though
you're only running your dialplan on one of them. Luckily the dialplans don't 
care
which incoming SIP line rings when Google Voice tries to call you. Therefore, by
forcing SIP Sorcery to attach to sipgate using both of the SIP Sorcery servers, 
the
SIP Sorcery server running your script is guaranteed to receive the call since 
it
will appear on both SIP Sorcery servers at the same time.

Original comment by easter...@gmail.com on 20 Jan 2010 at 1:13

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Weird.. I did both of those and changed my dial plan to the simple one again 
and made
sure all of the settings were correct.  It was working "OK" last night (still 
had
some issues), and this morning, it doesn't work at all (outbound calling that 
is). 
Inbound calling also doesn't even work on it.  I don't get what keeps 
happening?  Any
ideas at all?

Original comment by UCIR...@gmail.com on 20 Jan 2010 at 5:04

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Hi, easter...@gmail.com.

I am testing your method, but kind of confused.

When I use GV+SIPSORCERY (both gizimo5 and sipgate as SIP provider), more than 
50% of
chances I couldn't place the call successfully, either "temporarily not 
available" or
"host sipgate unresolvable" or "busy here". I am using sprint data connection. 
Is
this intrinsic problem of SIP, sipsorcery settings, or sprint network 
connection?

I set both GIZIMO5 and sipgate in the sipsorcery with the complex dial plan. 
What is
the advantages of setting both SIP providers there, over only set gizimo5 or 
sipgate?
 I am using Expresstalk on my windows mobile phone, which could set two lines at same
time. So I set up both sipsorcery(with GIZIMO5 and sipgate as sip provider) and
Gizimo5 as SIP. I found if I dial out with the line using Gizimo5 as SIP 
provider, it
is always much easier to get connected, compared with the line using sipsorcery 
as
SIP provider. Is there a possible reason under this?

Third, I am also trying to use NIMBUZZ on my cell phone with identical setting 
as I
use ExpressTALK or FRING. But with NIMBUZZ, there was no sound after connection 
is
established. Do you know what is possible solution?

Sorry ask too much here. Still have lots of questions!
Thanks!

Original comment by yongx...@gmail.com on 21 Jan 2010 at 1:00

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I have stumbled upon a solution: 
1) set up two different VOIP phones at Sipgate.com (or your voip provider) 
2) Set up an incoming calls dialplan. I named mine "IN". Like this:

if sys.IsAvailable() then
  sys.Dial( "MYSIPSORCERYUSERNAME@local[tr=b]")
end

3) See this for more info about transfers:
http://sipsorcery.wordpress.com/2010/01/03/transfers/

Original comment by t.hook...@gmail.com on 26 Jan 2010 at 1:15

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Anyone else having the issue where you dial out and you immediately get a fast 
busy
signal?

Original comment by UCIR...@gmail.com on 11 Feb 2010 at 8:11

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I've been noticing some problems lately with dialing through SIPSorcery.

I have 100% working outbound and incoming dialing, but when I do dial out, 
there is 
some amount of word loss / static in the line.

I've tried calling my state's Chase Bank weather phone (reports on the current 
temperature and time outside), which almost usually never has any cutouts or 
word 
loss (due to being a robot on the other end)...Except today. Today, it was 
cutting 
out every couple of words or skipping on parts of words.

I also was in another call earlier and I actually got disconnected while on the 
phone 
with them (they called back my Google Voice number, and we finished talking, 
without 
the slightest disconnect at all).

I seem to sound better and hear better the other person while using SIPSorcery 
/ 
SIPGate to dial out, but I suffered that disconnect earlier while we were 
talking, 
and then when they called back, we had some static / word cutoff, but just a 
little 
bit more.

Maybe Google or SIPSorcery or SIPGate is having some issues today with network 
usage 
or something, seeing as this (word cutoff, or random disconnects, or static) 
has 
never happened before.

I do have my router properly configured to port forward to my laptop currently 
logged 
in (ports 5000, and 8000-20000), and I usually never see this kind of stuff 
(SIP 
dialing's quality is almost always equal to my phone carriers quality).

Original comment by XANAVi...@gmail.com on 14 Feb 2010 at 3:11

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I forgot to put:

This started right about the time I was forced to remove Kaspersky AV and 
replace it 
with another antivirus (due to Kaspersky's beta license being removed as the 
beta 
program is now stopped, and so I can no longer use it).

I've currently got Avast! AV v5. Is there anyone else out there with this same 
setup 
who can say it is not just Avast! causing dropped packets?

I'd really like to avoid switched antiviruses again if I don't need to (as any 
other 
AV with a network scanning ability might mess it up still).

Original comment by XANAVi...@gmail.com on 15 Feb 2010 at 6:03

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Anyone having dial out problem? I setup everything couple days ago and used to 
work
fine. Today somehow getting busy tone after about 30 sec. of dialing out.
current setup: pap2t+sipsorcery+gizmo+GV

Original comment by azha...@gmail.com on 17 Feb 2010 at 2:00

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Hey, EasternPA.
I checked on the website www.robtex.com about sipsorcery's 2 servers... It 
seems that  
sip1.sipsorcery.com points to 174.129.236.7, and sip2.sipsorcery.com points to 
174.129.234.254.

If you switch over the registry contacts so they point to the new hostnames, 
you can 
avoid an issue where SIPSorcery might sometime in the future change those IP 
Addresses and break SIP connectivity if you are registered against those 
addresses 
and they change.

Plus, it is much easier to remember, and if SIPSorcery adds a new server, it 
will 
most likely be named sip3.sipsorcery.com, and will be much easier to add in 
than 
trying to find the address of the (currently nonexistent) 3rd server and 
registering 
against that.

Original comment by XANAVi...@gmail.com on 17 Feb 2010 at 4:16

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Thanks for checking in with the FQDNs, XANAVirus. Is anyone on this ticket still
having issues?

Original comment by easter...@gmail.com on 18 Feb 2010 at 12:15

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I am getting unstable dialing using fring. i have setup sipgate with 2 phones 
as 
suggested and also added the 2 SIP Providers in sipsorcery. Dialing with fring, 
i get the 
fake ring sound that fring has and then it disconnects saying call terminated. 
I am using 
the Simple Dialing plan.

Original comment by thienpha...@gmail.com on 16 Mar 2010 at 3:35

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why is this so inconsistent using fring? I am able to call via softphone on my 
laptop but 
not using fring on my iphone. It works very inconsistently with fring, 
sometimes it will 
put the call through,  other times it just terminates the call immediately, and 
other 
times it will just ring forever!

Original comment by thienpha...@gmail.com on 16 Mar 2010 at 4:43

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Sometimes this happens with my Fring, too. Now that I changed all registrations 
from 
sipsorcery.com to sip1.sipsorcery.com it's much more stable but still, I 
suggest that 
you use a "native" SIP client rather than go thru yet another server (fring).

Original comment by mte...@gmail.com on 24 Mar 2010 at 7:56

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Yea I actually just got a Nexus One and set it up using Sipdroid and it has 
been working pretty consistently. So I 
am using Sipdroid > Sipgate > Sipsorcery > Google Voice. Its been working 
pretty good the last couple days with 
both inbound and outbound calls. My only small gripe is that the audio sound a 
little robotic if that makes any 
sense. I think the issue is with sipdroid because I am using the same setup on 
my mac laptop using Telephone 
0.14.3 and the sound is better, more natural sounding and less robotic.

Original comment by thienpha...@gmail.com on 27 Mar 2010 at 1:05

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Still having the same issue now.  If I make an outgoing call from my VOIP line, 
it
rings and rings and then goes to a busy tone (even though the other end can 
pick up
the line, but nobody is there).

Anyone else having this issue?

Original comment by UCIR...@gmail.com on 31 Mar 2010 at 4:40

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I've had rings turn to busy, but not when the line I'm calling is not actually 
busy.
Remembering the first set of rings are the "fake" rings designed to make you 
think
the call is being placed, but the second set of outbound tones (either ring or 
busy)
is typically the actual outbound indicator coming from Google Voice. Do you 
test with
the podlinez numbers?

Also, if you go to podlinez, the original intent was to connect those podcasts 
to
real phone numbers first, so each podcast has a traditional number associated 
with
it. Dialing the real numbers associated with the podcasts is a much better 
end-to-end
test of your setup than dialing the SIP versions. Not that there's much I can 
really
suggest to correct the issue, but I would like to know how frequently calls 
placed to
those numbers actually go through.

It would be great if someone could programmatically test the call completion 
rates
and assess the audio quality.

Original comment by easter...@gmail.com on 31 Mar 2010 at 6:07

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arggg....it was working pretty good the last couple days but today. It keeps 
ringing forever again. Logged onto 
sipsorcery and checked under calls. It says Status 480, Ans. Reason Temporarily 
Unavailable. Been trying for the 
last hour or so!!

Original comment by thienpha...@gmail.com on 31 Mar 2010 at 9:07

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Works better for me now.. Went into sipsorcery and realized my sipgate phones 
were
unregistered.  Check register and now we're good!

Original comment by UCIR...@gmail.com on 31 Mar 2010 at 9:10

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yea i just did that too...it works now. I also changed my server from 
sipsorcery.com to sip1.sipsorcery.com

Original comment by thienpha...@gmail.com on 31 Mar 2010 at 9:28

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Okay good. I will leave this open for a short while longer. Gotta get this 
issue list
trimmed down a bit.

Original comment by easter...@gmail.com on 31 Mar 2010 at 9:30

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@thienpham.us

Why did you change your server to sip1.sipsorcery.com? Is it because the sip1 
is 
more reliable than sip?

Original comment by victor...@gmail.com on 1 Apr 2010 at 1:42

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after logging into sipsorcery.com its under SIP Providers. Click on your sip 
provider which you can then edit the 
settings. Check Register and for Register Contact this is where I changed it to 
sip1.sipsorcery.com. Also most 
software telephone asks you to provide account information like username/pass 
and domain, so I changed the 
domain there also. It seems pretty reliable for me. Only issue is when I am not 
on WIFI and using 3G from AT&T, 
there's a lag before the person can here my voice.

Original comment by thienpha...@gmail.com on 1 Apr 2010 at 1:51

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Sorry I misread your post. I changed it based on recommendation from mtelis in 
the post above. Seems pretty 
stable when using WIFI with good connection. Only issue i have is 3G there a 
slight delay in audio.

Original comment by thienpha...@gmail.com on 1 Apr 2010 at 1:55

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There are two Sipsorcery servers; sip1.sipsorcery.com points to the first one 
and 
sip2.sipsorcery.com -> to the second.

Both sipsorcery.com and sip.sipsorcery.com use quite tricky switching mechanism 
intended for load balancing and failover switching between the two servers. 
Unfortunately, this mechanism doesn't work well, especially if you need to 
receive an 
incoming call "forwarded to" your account @sipsorcery. Let me explain this in 
detail.

There are two ways you can get incoming calls. First, you can register with DID 
provider (check "Register" box in your Sipsorcery "SIP providers" settings for 
this 
provider). Second, you can go to the DID provider's configuration page and set 
it to 
forward incoming calls to your Sipsorcery SIP account. Sometimes the latter is 
the 
only way (in particular, this is the case with IPKall.com, you can't register 
to your 
IPKall DID).

Why would you care about calls if you only need to make calls using Google 
Voice? 
That's because Google Voice works via callback. If callback didn't reach your 
account, you'd observe a behavior described in EasternPA's post above: fake 
ringing 
while waiting for the callback turning into busy upon callback time-out.

When you configure your ATA or softphone to use sip1.sipsorcery.com (or 
sip2.sipsorcery.com), you disable switching mechanism. Of course, your DIDs 
must 
forward incoming calls to the same server (either sip1 or sip2), you should do 
it in 
their configuration page. If you "register into" provider, make sure your 
"Register 
Contact" is in "yourname@sip1.sipsorcery.com" format (as opposed to 
yourname@sipsorcery.com").

Original comment by mte...@gmail.com on 1 Apr 2010 at 5:06

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Original comment by easter...@gmail.com on 28 Apr 2010 at 6:00

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