Step by step for GV and Sipsocery help

mjc8870 · Post by **mjc8870** » Mon Aug 31, 2009 12:14 am

Can someone get me started so I can use this setup. I am completely new to this and have no knowledge of ruby code. I am looking to use GV and SS with ATA without running computer. I already have a gizmo account as I understand this is needed also.

Thanks for the help.

macman4hire · Post by **macman4hire** » Mon Aug 31, 2009 1:26 am

Unless you use a local version of Sip Sorcery, you will have wait until Sept. 8,2009, as Sip Sorcery has stopped excepting new accounts at this time.

mjc8870 · Post by **mjc8870** » Mon Aug 31, 2009 11:46 am

I already have sipsocery account.

Cablespider · Post by **Cablespider** » Mon Aug 31, 2009 11:16 pm

http://www.phoneservicesupport.com/goog ... t7299.html

synchron · Post by **synchron** » Tue Sep 01, 2009 12:54 pm

Since last Friday, any free incoming provider can be used w/ sipsorcery and I'm going on the 4th day now using Sipgate (replacing Gizmo) and it's been rock solid. I recommend using them for your GV callback a la sipsorcery.

GV/GG/SS is my primary ATA method now, mjmd5/ATA is my backup in case SS goes down, which is pretty rare these days.

Synchron

macman4hire · Post by **macman4hire** » Tue Sep 01, 2009 3:44 pm

hi synchron,

Please post your dial plan that your are using with your Google Voice, Sipgate and Sip Sorcery account.

Thanks,
macman4hire

synchron · Post by **synchron** » Tue Sep 01, 2009 11:01 pm

Sure thing, it's a little messy but things are commented out in case there's trouble. So far, no trouble. This is an Out dial plan only, no In necessary:

Code: Select all


#Ruby
# Dial Plan Generated by Rubyzard v0.1
# If you need help, please post in our forum
# http://www.mysipswitch.com

# SIP tracing : true or false
sys.Trace = true

sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")

if sys.In then
  # Do your INCOMING call processing customisations here.
    if sys.IsAvailable() then
       sys.Dial("#{sys.Username}@local",30)
       #sys.Dial("Enter Number@Gizmo5",30)
        #sys.Dial("Enter Number@sipgate",30)

       sys.Respond(480, "#{sys.Username} Not available")
    else
       #sys.Dial("Enter Number@Gizmo5",30)
        #sys.Dial("Enter Number@sipgate",30)
       sys.Respond(480, "#{sys.Username} Not available")
    end
 
else
  # Do your OUTGOING call processing customisations here.
    case req.URI.User
       when /^(\+?1)?(8(00|55|66|77|88)[2-9]\d{6})/ then sys.Dial("sipgate") 
       #when /^[1-9]/ then sys.Dial("LocalPhone")
       #when /^[1-9]/ then sys.GoogleVoiceCall("[email protected]", "GV Password", "Gizmo5number(1747XXXXXXX)", "#{req.URI.user}")
       when /^[1-9]/ then sys.GoogleVoiceCall("[email protected]", "GV Password", "Sipgate Number", "#{req.URI.user}", "GV Number")
       #when /^[1-9]/ then sys.GoogleVoiceCall("[email protected]", "GV Password", "Voxox Number", "#{req.URI.user}", "GV number")
       else sys.Dial("LocalPhone")
    end
 
end

Synchron

anant · Post by **anant** » Thu Sep 03, 2009 8:44 pm

Hi Synchron:

I just modified 3rd line from bottom in your dial plan as follows. That is the only change I made in your dial plan for sipgate.

when /^[1-9]/ then sys.GoogleVoiceCall("[email protected]", "GV pw", "Sipgate Number", "#{1415XXXYYYY}", "1440xxxyyyy")

Is there any other place in the dial plan I should make changes ?

I am getting dial tone alright. But getting a busy tone when I dial a number.

This is my problem right now. I suspect I have a problem in dial plan. Do you see problem anywhere else?

I have PAP2T where I entered sipsorcery data.

I need your help. Thanks in anticipation.

anant

anant · Post by **anant** » Thu Sep 03, 2009 8:50 pm

Hi Synchron:

Incoming calls going alright. I have problems with Outgoing calls. Getting dial tone. Appreciate any help.

anant · Post by **anant** » Thu Sep 03, 2009 8:57 pm

What is

"#{req.URI.user}" I filled in the sipgate telephone I got.

"#{1415XXXYYYY}" Is it correct or am I missing something ?

anant · Post by **anant** » Thu Sep 03, 2009 10:16 pm

I would like to check my Sipgate settings in SS.
Need Help. getting busy tone for out calls Inc. calls ok.

Here are mine:

Provider name: sipgate
username: xxxxxxxxxx as given by sipgate
password: zzzzzzzzzz as given by sipgate
server: sip:sipgate.com:5060
Register checked
Register contact: sip:[email protected]

Outbound proxy: sip:sipgate.com
Authorized user name: username: xxxxxxxxxx as above
From Header: blank
Reg. Expiry: 3600
Reg. Realm: sipgate.com
Reg. server: sip:sipgate.com:5060
Custom Headers : blank

Note: I am not using the telephone# I got from sipgate 1415xxxyyyy.

Dial Plan:

#Ruby
# Dial Plan Generated by Rubyzard v0.1
# If you need help, please post in our forum
# http://www.mysipswitch.com

# SIP tracing : true or false
sys.Trace = true

sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")

if sys.In then
# Do your INCOMING call processing customisations here.
if sys.IsAvailable() then
sys.Dial("#{sys.Username}@local",30)
#sys.Dial("Enter Number@Gizmo5",30)
#sys.Dial("Enter Number@sipgate",30)

sys.Respond(480, "#{sys.Username} Not available")
else
#sys.Dial("Enter Number@Gizmo5",30)
#sys.Dial("Enter Number@sipgate",30)
sys.Respond(480, "#{sys.Username} Not available")
end

else
# Do your OUTGOING call processing customisations here.
case req.URI.User
when /^(\+?1)?(8(00|55|66|77|88)[2-9]\d{6})/ then sys.Dial("sipgate")
#when /^[1-9]/ then sys.Dial("LocalPhone")
#when /^[1-9]/ then sys.GoogleVoiceCall("[email protected]", "GV Password", "Gizmo5number(1747XXXXXXX)", "#{req.URI.user}")
when /^[1-9]/ then sys.GoogleVoiceCall("[email protected]", "myGVpw", "Sipgate Number", "#{1415xxxyyyy}", "1440xxxzzzz")
#when /^[1-9]/ then sys.GoogleVoiceCall("[email protected]", "GV Password", "Voxox Number", "#{req.URI.user}", "GV number")
else sys.Dial("LocalPhone")
end

end

Josemiami · Post by **Josemiami** » Thu Sep 03, 2009 10:28 pm

Sample of new Aaron Logic for Google Voice Outbound Calling Using Sipgate:

Code: Select all

####################################################
######## OUTBOUND CALL FROM GOOGLE VOICE  ##########
####################################################

#Logic for routing outgoing calls.
sys.Trace = false
case req.URI.User
when /^/
   sys.Log(" Dialing USA #{req.URI.User} through Google Voice service\t")
   sys.Log("*****************************************************\n")
   sys.GoogleVoiceCall("[email protected]","GV_Password","415XXXXX19","#{req.URI.User}","786XXXXX86")
end

[email protected] = is your GV email account.
GV_Password = GV account Password
415XXXXX19 = Sipgate Number with or without the "1", This number have to be registered with SS, and have to be a forwarding number on the GV account.
786XXXXX86 = This is your GV number you want to match in this call, and for this set. make sure this number does not have a 1 in front or it won't work. (your GV email account and Password are the ones used for this Number).
#{req.URI.User} = any number you will call using SS, leaved as it or specified a particular number.

You can get your Sip credentils from your sipgate account it will look like this:

Add your Sipgate Number as a forwarding number on your Google voice Account:

This is all you need you dont need the IN Dial Plan

mel2000 · Post by **mel2000** » Thu Sep 03, 2009 10:57 pm

anant wrote:What is

"#{req.URI.user}" I filled in the sipgate telephone I got.

"#{1415XXXYYYY}" Is it correct or am I missing something ?

Only variables converted to strings need to be enclosed in braces. "1415XXXYYYY" is valid syntax, so is "#{req.URI.user}", but not "#{1415XXXYYYY}".

Josemiami · Post by **Josemiami** » Thu Sep 03, 2009 11:00 pm

"#{req.URI.user}" don't put anything in here leave like that.

anant · Post by **anant** » Thu Sep 03, 2009 11:01 pm

Hi Jose:

Thx for your reply.

I do not understand

415XXXXX19 = Sipgate Number with or without the "1", This number have to be registered with SS and have to be a forwarding number on the GV account.

Where does this number go in SS. ?

I had created sipgate account under GV and have the # 1415 xxxx as a fwd. #.

Is there no need for filling in Outboung proxy , authorized id etc under SS sipgate provider ?

I would try your dial plan after fixing the above.

anant · Post by **anant** » Thu Sep 03, 2009 11:34 pm

Hi Jose:

I seem to be doing everything right as per your screenshots. I copied your dial plan. But still I have busy tone for outbound.

I need answers to my earlier post. Somehow the sipgate tel # does not find a place in SS. I do not see it also in your screen shots. That seems to be the problem for me now.
You also did not fill in Outboundproxy / authorization id etc

Kindly clarify the above. Thanks in anticipation. I seem to be close but not there yet.

Josemiami · Post by **Josemiami** » Thu Sep 03, 2009 11:44 pm

In order to register your sipgate Number with Sipsorcery you have to get the sip credentials from your sipgate page and register under sip providers on sipsorcere as in the pics.
If you look at the pics you are going to see there is a sipgate entry as one of the sip providers this is the way you register the number with sipsorcery.
Just look at the pics.

anant · Post by **anant** » Fri Sep 04, 2009 12:06 am

I have it as follows under SIP providers.

provider owner user password server register
sipgate mine xxxxe1 yyyyyyy sipgate.com chk

I do not have 1415xxxxxx associated with the above directly

We are on the same page.

What about Outbound proxy , authorized id etc under SIP providers data for sipgate ? Since you did not show may be there is no need to fill.

But still why am not getting my OUT call thru ? Everything appear to be matching with your screens .

That is my big problem.

Josemiami · Post by **Josemiami** » Fri Sep 04, 2009 12:09 am

Code: Select all

What about Outbound proxy , authorized id etc under SIP providers data for sipgate ? Since you did not show may be there is no need to fill.

Yes no need to fill.

anant · Post by **anant** » Fri Sep 04, 2009 12:20 am

You have answered all my questions.

Is there anything else you want me to look at?

It appears to me that the sipgate tel # 1415xxxYYYY does not have to be entered anywhere else except in the dial plan. Of course under GV account.

Everything seems to match with your screens but still busy tone after the OUT call.

Probably I will take another look tomorrow. It is mysterious. If you think of anything else I should take a look please let me know. Thanks for your time.

Josemiami · Post by **Josemiami** » Fri Sep 04, 2009 12:21 am

Did you added your Dial Plan to your SS account?

Josemiami · Post by **Josemiami** » Fri Sep 04, 2009 12:25 am

anant Just be patient and double check every thing I think you are in the right track.

anant · Post by **anant** » Fri Sep 04, 2009 12:38 am

Yes I have dialout plan as sipgate under SIP Accounts. It matches with your screens.

Josemiami · Post by **Josemiami** » Fri Sep 04, 2009 12:56 am

Code: Select all

sys.GoogleVoiceCall("[email protected]","GV_Password","415XXXXX19","#{req.URI.User}","786XXXXX86")

Anant, the last number is suppose to be your GV Number and as I said before it cant have a 1 in front or it wont work.

anant · Post by **anant** » Fri Sep 04, 2009 1:02 am

It does not have a 1 as suggested by you earlier.

desidude2000 · Post by **desidude2000** » Sat Sep 05, 2009 3:24 am

anant wrote:It does not have a 1 as suggested by you earlier.

anant, please post the dial plan you're using... i was having problems with a complicated dialplan, but simple dialplan that jose posted worked for me. the complicated dialplan (with handling for incoming) gave me a fast busy signal.

anant · Post by **anant** » Sat Sep 05, 2009 7:49 am

Hi:
This is my dial plan for sipgate in SS. This is also quite simple.

-----------------------------------------------------------------------------------
####################################################
######## OUTBOUND CALL FROM GOOGLE VOICE ##########
####################################################

#Logic for routing outgoing calls.
sys.Trace = false
case req.URI.User
when /^/
sys.Log(" Dialing USA #{req.URI.User} through Google Voice service\t")
sys.Log("*****************************************************\n")
sys.GoogleVoiceCall("[email protected]","my pw","415xxxyyyy","#{req.URI.User}","my 10-digit GV # ")
end
--------------------------------------------------------------------------------
I got my problem solved receiving continued help from Josemiami. The busy tone just after dialing a number disappeared with the following 2 changes in PAP2T.
I always thought that there is no need to change the dial plan in PAP2T. I used to have a dial plan when I was using GVOUT containing the computer IP address etc. I replaced that with a regular dial plan which is working on my other line of PAP2T. That is one change I made.
The second change was: I reduced the Register expiry from 600 to 180.

After making the above 2 changes, I am able to make out calls and no more busy tones.

Thanks for your response.

anant · Post by **anant** » Sat Sep 05, 2009 12:26 pm

Hi folks:

Finally I got sipgate and voxox working with SS for both out/in calls.

I have still problem with Gizmo:

When I call an outside # from GV contact list , my phone rings and when I pick up it rings and make the call. and the call goes thru. No problem

But when I make a call from regular phone connected to PAP2T. after 3 rings I get a busy tone. I do not know what is going on. I need help if someone can jump into.

Here are my settings ( as I gathered from this thread):

I have gizmo # 1747xxxyyyy under GV account

Under Gizmo5 account:
I have GV ON and have GV # so that I can qualify for a free 3 min call.
I have deactivated Voice mail
Call forwarding to [email protected]

In Sipsorcery :
In SIP Accounts I have Gizmo dial plan

In SIP Providers:
Provider name Gizmo5
Username: 1747xxxyyyy
password: my password
server: sip:proxy01.sipphone.com
Register: check mark
Register contact : sip:[email protected]

OUT Dial Plan for Gizmo: A very simple plan

####################################################
######## OUTBOUND CALL FROM GOOGLE VOICE ##########
####################################################

#Logic for routing outgoing calls.
sys.Trace = false
case req.URI.User
when /^/
sys.Log(" Dialing USA #{req.URI.User} through Google Voice service\t")
sys.Log("*****************************************************\n")
sys.GoogleVoiceCall("[email protected]","mypw","747xxxyyyy","#{req.URI.User}")
sys.Log("Sorry, Google Voice Call failed.")
end

I have register expiry at 180.

______________________________________________

It is not obvious to me what I am doing wrong. But when I make a call from regular phone connected to PAP2T. after 3 rings I get a busy tone. I do not know what is going on. I need help if someone can jump into.
Thanks in anticipation.

synchron · Post by **synchron** » Sun Sep 06, 2009 3:16 am

If you use the GV callback, you don't need to enable GV in your G5 account for the free 3 minute thing, it will work unlimited because it is considered incoming to G5 service. Also, u might try using "1747" for the G5 number. And make sure the G5 has been added and checked in your phone setup in GV.

As you get used to using SS, you can try using the console to troubleshoot your problems, especially the fast busys you get.

Regards,

Synchron

MJuser403 · Post by **MJuser403** » Sun Sep 06, 2009 9:35 am

synchron wrote:If you use the GV callback, you don't need to enable GV in your G5 account for the free 3 minute thing, it will work unlimited because it is considered incoming to G5 service. Also, u might try using "1747" for the G5 number. And make sure the G5 has been added and checked in your phone setup in GV.

As you get used to using SS, you can try using the console to troubleshoot your problems, especially the fast busys you get.

Regards,

Synchron

Do you need Gizmo any more if you get sipgate + google voice + sipsorcery?

I guess I'm asking, is the sipgate+google voice+sipsorcery totally free incoming and outgoing with no limits?

anant · Post by **anant** » Sun Sep 06, 2009 9:51 am

As far I could see with my limited testing, sipgate is working for both in/out calls. Just wondering why I am not able to make Gizmo also work in case there is any need to fall back on. Besides, Gizmo worked for all on this board and why not me is the question.

synchron · Post by **synchron** » Mon Sep 07, 2009 9:14 am

Sipsorcery server was very sporadic yesterday - I woke up with no voip light so ATA was not registered until like 4 hours later where it came back online and then went offline again not more than 1 half hour later.

I left the house after it was back online but when I came back, it was offline again. Woke up today and it was back online. It is still considered an "experimental" service and at least I can use my Mj as a backup and just switch the phone cord.

Interestingly enough, yesterday's SS outages were unlike ones in the past where whenever the ATA was not registered, the website was down too. This time the website was running yet no ATA could link to the SS servers and I also learned that pinging the server never gives you any packets so Aaron, leader of SS, is really doing some IP magic with his Voip switching system. Perhaps, he was doing some maintenance yesterday, maybe even expanding the servers to handle more throughput in hopes of allowing new accounts in a few days.

Hopefully, the kinks are worked out because when SS is working, I can now do GV callbacks (free US calls treated as free incoming) now with Sipgate, VoxOx, Gizmo 5, and more recently, IPKall (which is crystal clear reception).

[edit - Aaron just recently responded that some idiot was hammering the SS registrar, thus, the reason for the service going haywire yesterday. Unfortunately, since this is always possible, the 9/8 date has been removed and no new accounts are allowed. I've asked to lend a hand but if anyone has C# programming/development skills so that SS becomes more stable, perhaps at some later time, new accounts will be allowed - http://www.mysipswitch.com/forum/index.php - leave word for Aaron if interested].

Regards,

Synchron

synchron · Post by **synchron** » Mon Sep 07, 2009 9:23 am

MJuser403 wrote: Do you need Gizmo any more if you get sipgate + google voice + sipsorcery?

I guess I'm asking, is the sipgate+google voice+sipsorcery totally free incoming and outgoing with no limits?

Yes to sipgate but this is the one provider that won't let you do Toll-free without registering for a DID. There's a slew of providers that allow toll-free directly without the need to register them to get a DID. They seemlessly integrate to the SS dialplan.

Synchron

synchron · Post by **synchron** » Mon Sep 14, 2009 12:55 pm

Just to let every1 know, there is currently a window of opportunity to make an SS account for the next 2 to 3 days.

Read here first: http://sipsorcery.wordpress.com/

This is a great alternative to using your ATA/mobile SIP and dialing USA/Canada for free via Google Voice Callback and an umpty dum # of SIP providers that give you free incoming service. Your MJ can get a "rest" and can be there for a backup.

Synchron

Cybernaut · Post by **Cybernaut** » Mon Sep 28, 2009 1:39 am

Synchron or anyone else, can you post the PAP2 settings for use with SIPSorcery/Google Voice/SIPGate. I think I've got everything else figured out. Thanks.

Cybernaut · Post by **Cybernaut** » Thu Oct 01, 2009 10:41 pm

Cybernaut wrote:Synchron or anyone else, can you post the PAP2 settings for use with SIPSorcery/Google Voice/SIPGate. I think I've got everything else figured out. Thanks.

Bump. Anyone?

synchron · Post by **synchron** » Fri Oct 02, 2009 6:32 pm

Sorry, don't own a PAP2 but minimum general ATA or mobile apps like Fring, Siphon, Isip, Ipico, etc. SIP credentials for sipsorcery are:

Username = your Sipsorcery SIP account username
Password = your Sipsorcery SIP account password
domain/proxy server: sipsorcery.com
Reg. Timeout = 3600 or 1800 (recommended)

You can test that your ATA/mobile is recognized by sipsorcery by logging into your Silverlight capable browser and checking to see that your IP address in SS account under Contact URI is bounded - click refresh if necessary. You can even see your ATA brand-model or your Sip mobile application under the user-agent string field to the right of proxy socket.

Synchron

anant · Post by **anant** » Fri Oct 02, 2009 6:40 pm

This link could be helpful.

http://google-voice-tricks.the-up.com/g ... ate-t7.htm

anant

VoipDude · Post by **VoipDude** » Sat Oct 03, 2009 8:04 pm

I have been using SipSorcery with an ATA successfully for outgoing through GV, and incoming through IPKall.

Today, this stopped working. SipSorcery is registered fine, and my ATA is registered to SipSorcery, but it appears something changed with Google Voice that it ceased working.

Is anyone else experiencing this, and if so, have you been able to correct it?

anant · Post by **anant** » Sat Oct 03, 2009 8:17 pm

Yes. IPKALL not working since 5:00 AM Est this morn when I started noticing. A busy tone after 2 rings. Even GV not able to call. It seems to be the longest outage. Please see some others also complaining about this on IPKALL forum. I think IPKALL must be going thru some upgrade. Others like sipgate/gizmo/voxox are all working with SS except IPKALL. Let us be patient. Afterall it is free sweet service. We must put up with some inconvenience. Let us hope for the best.

anant

VoipDude · Post by **VoipDude** » Sat Oct 03, 2009 8:24 pm

anant wrote:Yes. IPKALL not working since 5:00 AM Est this morn when I started noticing. A busy tone after 2 rings. Even GV not able to call. It seems to be the longest outage. Please see some others also complaining about this on IPKALL forum. I think IPKALL must be going thru some upgrade. Others like sipgate/gizmo/voxox are all working with SS except IPKALL. Let us be patient. Afterall it is free sweet service. We must put up with some inconvenience. Let us hope for the best.

anant

Thanks anant.

Even though it's frustrating, it's good to know that the problem is with IPKall. This way I'm not barking up the wrong tree. Hopefully IPKall will get things up and running soon. I really can't complain about free service Thanks for the encouragement.

synchron · Post by **synchron** » Mon Oct 05, 2009 1:27 pm

Some other folk at the MSS forums have reported that IPKall is working again but for me, it still don't work (since Sat Morning). Wait, I forget, do I have to check it in GV settings again for the callback to work?

For that matter, VoxOx has been spittiing 480 errors at me - only Sipgate and Gizmo5 have been nice to me.

Synchron

anant · Post by **anant** » Mon Oct 05, 2009 3:54 pm

Hi Synchron:

checked all by changing the dial plans in SS one by one just now. All ( gizmo. IPKALL, sipgate, and Voxox) are working. Sometimes we need to try a second time.

anant

synchron · Post by **synchron** » Mon Oct 05, 2009 8:18 pm

Yeah, I think GV callback only needs the SS dialplan logic in the outgoing section. You only need to check/forward number in GV phone settings to match what you check for your Contact URI in your SS account settings for incoming phone calls.

I'll try again tonight but as of this morning, VoxOx and IPKall were still not working for me.

Edit [10/06]: OK IPKall working again. Definitely the best provider as far as VQ and fastest callback. Too bad it only rings the ATA, not my Ipod Touch using iSip, iPico, Siphon or Fring.

Sipgate is 2nd - decent VQ and callbacks take 6-8s (kinda close to the SS limit)

Gizmo5 - 3rd, worst as far as VQ

VoxOx - still don't work. Sometimes I can't even get the softphone to ring.

Synchron

VoipDude · Post by **VoipDude** » Wed Oct 07, 2009 10:59 am

ATA down again this morning.

Not sure if it's SipSorcery or IPKall.

Worked fine yesterday....no changes made on my end.

Anyone else having problems?

anant · Post by **anant** » Wed Oct 07, 2009 12:45 pm

I have tested in the morn and just tested now after seeing your msg. SS + IPKALL working well both for in/out calls.

anant

VoipDude · Post by **VoipDude** » Wed Oct 07, 2009 2:11 pm

After rebooting both my modem/router & ATA IPKall is working again, however, there is no audio on incoming calls. Works fine for outgoing calls.

The issue of no audio on incoming calls is a current issue being discussed in the MySipSwitch forum. Aaron is aware of it, but at the present is not sure how to correct it.

The link for that forum is below:

http://www.mysipswitch.com/forum/viewtopic.php?t=1781

VoipDude · Post by **VoipDude** » Wed Oct 07, 2009 2:12 pm

anant wrote:I have tested in the morn and just tested now after seeing your msg. SS + IPKALL working well both for in/out calls.

anant

Thanks for the status report. It's always helpful to see what others are experiencing.

synchron · Post by **synchron** » Wed Oct 07, 2009 8:19 pm

Thanks to great forums and the posting of siptraces/sip events, the problem was easily corrected and the incoming/no voice downtime was kept to a minimum.

It's supposedly fixed now.

Synchron

VoipDude · Post by **VoipDude** » Sun Oct 11, 2009 10:59 pm

Back from vacation, and all is working well now.

magicJack and magicJack Plus Support, Reviews, FAQs and Tricks

Step by step for GV and Sipsocery help

Step by step for GV and Sipsocery help

Need help on your dial plan for sipgate.

Need help on dialplan sipgate

Need Help. getting busy tone for out calls Inc. calls ok.

SS + PAP2T + sipgate.

Need help SS+PAP2+sipgate

Need help

Need help.

Need help

need help

Re: need help

A problem with SS+Gizmo+PAP2T

sipgate working with SS. Why not gizmo is my question.

SS+sipgate+PAP2T

SS + IPKALL has problem - a busy tone after 2 rings.

Re: SS + IPKALL has problem - a busy tone after 2 rings.

SS +

SS + IPKALL working well

Re: SS + IPKALL working well