Google Voice Dial Out using Sipsorcery (no computer needed)
Moderators: Bill Smith, Pilot
Google Voice Dial Out using Sipsorcery (no computer needed)
Hopefully this will help some others out to get inbound and outbound calling using Google Voice and Gizmo5.
1. Setup Google Voice account
2. Setup Gizmo5 Account
3. Add Gizmo5 to your phones in Google Voice
4. Set up an account at http://www.sipsorcery.com/
5. Add a SIP Account to connect your ATA or softphone
6. Add SIP Provider Gizmo5
7. Add dial plan (see http://www.phoneservicesupport.com/post46080.html#46080)
8. Change outgoing setting on your SIP account to dial plan created in step 7
Now you should be able to dial out using your Gizmo5 line
1. Setup Google Voice account
2. Setup Gizmo5 Account
3. Add Gizmo5 to your phones in Google Voice
4. Set up an account at http://www.sipsorcery.com/
5. Add a SIP Account to connect your ATA or softphone
6. Add SIP Provider Gizmo5
7. Add dial plan (see http://www.phoneservicesupport.com/post46080.html#46080)
8. Change outgoing setting on your SIP account to dial plan created in step 7
Now you should be able to dial out using your Gizmo5 line
Andy Rogers
email me: U0pFTi16dxTvDdmlaK0+8bg2sZT+DlHLd6Jc7vG3M9ixlSKvrQgccJjf7tU=
email me: U0pFTi16dxTvDdmlaK0+8bg2sZT+DlHLd6Jc7vG3M9ixlSKvrQgccJjf7tU=
Yes Sipsorcery is base on silverlight, so if you are working on Linux for now it is no way, just install sirverlight in your Windows computer.digitize wrote:I was going to give this a try but when I go the sipsorcery link I get a weird Microsoft window saying to install Silverlight. What I am supposed to do? Is this normal?
Good luck
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Guide_timothy
- magicJack Apprentice
- Posts: 21
- Joined: Mon Dec 15, 2008 9:22 pm
I believe Google Voice has a 3hr time limit. I could be wrong but I think I read that somewhere.
Joe
Joe
laserjobs wrote:This way makes calls out using Google Voice not Gizmo5 so there is no time limitGuide_timothy wrote:AM I missing something here ??? there is still that annoying 3 minute outbound limit right ???? Or did you guys figure a work around ???
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Guide_timothy
- magicJack Apprentice
- Posts: 21
- Joined: Mon Dec 15, 2008 9:22 pm
WOW I set everything up correctly on sipsorcery and did the special Dial out plan and set up my ATA device I called my friend and we talked for like 15 minutes
All you guys are awsome at this
Unfortunately I'm afriad its only a metter of time before Gizmo or goole voice blocks sipsorcery remember Magic Jack blocked them So lets enjoy it while it lasts
here is a picuture of the format and the SIP info i entered for the add sip provider part on sipsorcery

make sure on your ATA you put sip.sipsorcery.com port 5060 in all the proxy outbound register and all that then your good to go
and of course your sipsorcery user name and password only thing goes in your ATA on this setup is is your sipsorcery stuff you put your gizmo SIP settings when you loginto sipsorcery.com
make sure on your ATA you put sip.sipsorcery.com port 5060 in all the proxy outbound register and all that then your good to go
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Cablespider
- magicJack Apprentice
- Posts: 28
- Joined: Wed May 27, 2009 4:12 pm
I tried this but I'm as soon as I looked at my Gizmo history, I saw this:
I was calling my cell to test. Is this normal?
Code: Select all
Date (GMT) Number Rate/min Duration Price
12 August, 05:42 PM +1-219-xxx-xxxx $0.0190 00:01 $0.019
12 August, 05:41 PM +1-219-xxx-xxxx $0.0190 00:01 $0.019Cablespider wrote:I tried this but I'm as soon as I looked at my Gizmo history, I saw this:
I was calling my cell to test. Is this normal?Code: Select all
Date (GMT) Number Rate/min Duration Price 12 August, 05:42 PM +1-219-xxx-xxxx $0.0190 00:01 $0.019 12 August, 05:41 PM +1-219-xxx-xxxx $0.0190 00:01 $0.019

Your SS dialplan is misconfigured and you are sending your outbound calls to Gizmo5 instead of Google Voice.Cablespider wrote:I tried this but I'm as soon as I looked at my Gizmo history, I saw this:
I was calling my cell to test. Is this normal?Code: Select all
Date (GMT) Number Rate/min Duration Price 12 August, 05:42 PM +1-219-xxx-xxxx $0.0190 00:01 $0.019 12 August, 05:41 PM +1-219-xxx-xxxx $0.0190 00:01 $0.019
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Cablespider
- magicJack Apprentice
- Posts: 28
- Joined: Wed May 27, 2009 4:12 pm
Yup...the dialplan was jacked up. I used the simple plan from Aaaron's tut and I can call out. I verified the call change through SS and can see the GV #.
Now, I can dial out but not in. When calling my GV #, it goes to VM. Google history sees the call but the phone won't ring.
Thanks for the heads up on the dialplan.
Edit: I figured it out my dial in problem. When setting up the SIP Provider, the Register Contact defaulted to the owner name. The SIP Account name for my ATA is something else. Once I changed the RC to that, I could dial in fine. My bad.
Now, I can dial out but not in. When calling my GV #, it goes to VM. Google history sees the call but the phone won't ring.
Thanks for the heads up on the dialplan.
Edit: I figured it out my dial in problem. When setting up the SIP Provider, the Register Contact defaulted to the owner name. The SIP Account name for my ATA is something else. Once I changed the RC to that, I could dial in fine. My bad.
Last edited by Cablespider on Thu Aug 13, 2009 7:51 pm, edited 2 times in total.
I dont think I have my RTP300 Setup correctly to use Sipcorcery. for the proxy I tried sip.sipsorcery.com and it failed to register. I tried proxy01.sipphone.com and it registers but I think it is connecting direct to Gizmo service. I would like my ATA setup to use Line1 for MagicJack and Line2 for my Google Voice account.
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desidude2000
- MagicJack User
- Posts: 36
- Joined: Mon Sep 22, 2008 4:16 pm
ok, this works great for outgoing, but in order to achieve this on my pap2, i entered the sip credentials of sipsocery on line1 of my ATA. but what about incoming now? i used to have incoming on line1 of my ATA such that, it was
(so i was using gizmo5 credentials on my line1).
how do you get BOTH incoming AND outgoing working on one line using the combination of GV+gizmo+sipsocery on a PAP2 on a single line?
Code: Select all
google voice -> gizmo5 -> ATAhow do you get BOTH incoming AND outgoing working on one line using the combination of GV+gizmo+sipsocery on a PAP2 on a single line?
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agriffiths
- magicJack Apprentice
- Posts: 26
- Joined: Wed Apr 08, 2009 10:45 pm
Figured out incoming on PAP2. Register only one line directly with Sipsorcery and get both in and out calls on that line.
I only have the gizmo5 sip account registered at SC. No other trunks. Part of this might be unnecessary, but this is what is working...
Follow the instructions in the beginning of the thread.
After signing up to sipsorcery, go to the original Sip Account, the one with the same name as your login for sipsorcery, and click it to Edit.
Tick the box that says Keep Alives.
For Out Dial Plan, choose the Default Dial Plan
For In Dial Plan, choose the blank box, so it says nothing. Save.
Next, this is the part I'm not sure is necessary, I created a second Sip Account, set up the same way as 1st, just diff name and/or password. Same dial plans for both.
Finally, Dial Plan. You should have no other dial plans except Default. No other ins or outs.
Default Dial Plan should be:
sys.Log("starting dialplan...")
sys.GoogleVoiceCall("[email protected]", "Gizmo5password", "1747612xxxx", "#{req.URI.user}")
sys.Log("Sorry, Google Voice Call failed.")
Log in on ATA
set up very simply to register to sipsorcery, with Original Sip Account, the one that's the same as you sipsorcery login
Only changes I made were:
Proxy: sip.sipsorcery.com
Register Expires: 180
User ID: name of 1st SIP acct on Sipsorcery
Password: the pass you created for this sip acct on sc
Stun Server: stun.ekiga.net
I only have the gizmo5 sip account registered at SC. No other trunks. Part of this might be unnecessary, but this is what is working...
Follow the instructions in the beginning of the thread.
After signing up to sipsorcery, go to the original Sip Account, the one with the same name as your login for sipsorcery, and click it to Edit.
Tick the box that says Keep Alives.
For Out Dial Plan, choose the Default Dial Plan
For In Dial Plan, choose the blank box, so it says nothing. Save.
Next, this is the part I'm not sure is necessary, I created a second Sip Account, set up the same way as 1st, just diff name and/or password. Same dial plans for both.
Finally, Dial Plan. You should have no other dial plans except Default. No other ins or outs.
Default Dial Plan should be:
sys.Log("starting dialplan...")
sys.GoogleVoiceCall("[email protected]", "Gizmo5password", "1747612xxxx", "#{req.URI.user}")
sys.Log("Sorry, Google Voice Call failed.")
Log in on ATA
set up very simply to register to sipsorcery, with Original Sip Account, the one that's the same as you sipsorcery login
Only changes I made were:
Proxy: sip.sipsorcery.com
Register Expires: 180
User ID: name of 1st SIP acct on Sipsorcery
Password: the pass you created for this sip acct on sc
Stun Server: stun.ekiga.net
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newwonder01
- magicJack Apprentice
- Posts: 21
- Joined: Sat Feb 02, 2008 10:14 pm
Checking Voicemail
Can someone provide details on how to setup the Sipsorcery dial plan so that GV voicemail can be checked directly from the phone. I have the GV / Gizmo / Sipsorcery Deal running great on an ATA, but can't get into the voicemail when I dial my own GV number?
Code: Select all
###################################################
######## OUTBOUND CALL FROM GOOGLE VOICE ##########
###################################################
#Logic for routing outgoing calls.
sys.Trace = false
case req.URI.User
when /^619/ # Calling a 619 number in Ca will show my GV number
sys.Log(" Dialing USA #{req.URI.User} through Google Voice service\t") # for Coronado Ca!!
sys.Log("*****************************************************\n")
sys.GoogleVoiceCall("[email protected]","GV1_Pass","1747XXXXX06","1#{req.URI.User}")
when /^815/ # Calling a 815 number will show my Chicago Gv Number
sys.Log(" Dialing USA #{req.URI.User} through Google Voice service\t")
sys.Log("*****************************************************\n")
sys.GoogleVoiceCall("[email protected]","GV2_Pass","1747XXXXX00","1#{req.URI.User}")
when /^1(800|866|877|888|747)\d{7}/ # G5 allows free Toll-Free calls including
sys.Dial("#{req.URI.User.to_str[0,11]}@Gizmo5") # 747 (Dialing a 1+ number)
when /^(800|866|877|888|747)\d{7}/ # G5 allows free Toll-Free calls including
sys.Dial("1#{req.URI.User.to_str[0,10]}@Gizmo5") # 747(Dialing just 10 Dig)
when /^411/ # G5 allows free 411 calls
sys.Dial("411@Gizmo5")
when /^7865551212/ # My Main GV account Number (Calling this Number
sys.Log(" Dialing USA #{req.URI.User} through Google Voice service\t") # using a differentGV account will give me access to
sys.Log("*****************************************************\n") # my Voice Mail, just hit * + pin after the system answer.
sys.GoogleVoiceCall("GV3_UserName","GV3_Pass","1747XXXXX30","#{req.URI.User}")
when /^/ # This will be my main GV account it is the only account
sys.Log(" Dialing USA #{req.URI.User} through Google Voice service\t") # reg with SS. All the others G5 Accounts will be
sys.Log("*****************************************************\n") # forwarding to my SS Account.
sys.GoogleVoiceCall("[email protected]","Main_GV_Pass","1747XXXXX99","#{req.URI.User}")
else # This Part of the dial plan wont be neccesary
#Route Outgoing Via Dialer Length # if you only dial 10 Dig.
dialer_length = req.URI.User.Length.to_s
case dialer_length
when /^10/
sys.Log(" If 10 digits, add the 1 and dial provider.\t")
sys.Log("--------------------------------------------------------------------------\n")
sys.GoogleVoiceCall("[email protected]","Main_GV_Pass","1747XXXXX99","1#{req.URI.User}")
when /^11/
sys.Log(" If 11 digits, dial provider.\t")
sys.Log("--------------------------------------------------------------------------\n")
sys.GoogleVoiceCall("[email protected]","Main_GV_Pass","1747XXXXX99","#{req.URI.User}")
end
endVoicemail Calling
This still does not work.....when I call my Main GV Number user the GoogleVoiceCall fron another GV number it dials and I can here the beep which indicates an incoming call waiting signal. But I think there is a way to dial and get directly into the mailbox.
Re: Voicemail Calling
Create a rule so that when you dial your GV number it goes out using Gizmo5 (call limited to 3 min).drecha wrote:This still does not work.....when I call my Main GV Number user the GoogleVoiceCall fron another GV number it dials and I can here the beep which indicates an incoming call waiting signal. But I think there is a way to dial and get directly into the mailbox.
Actually Gizmo5 wont let you get access to your own Gizmo phone it is block.
the only way to do it is the way I have posted. Just use a different set of G5 + GV number .
You don't have to register this set with SS, and the call is free and unlimited because you can call any phone number, and this number just happen to be your own main GV number wish is register with SS. you can use the same SIP phone, Soft phone or PAP2T to dial. Because is going out using a different Set Observe in my dial plan I am using various GV numbers for different purposes , only your main GV number must be register with Sipsorcery.
In this case when you dial 7865551212 (my main GV number), Sipsoucery will use my other set of GV + G5 (not registered with SS, to call my number, once is connected the phone will ring 4 times and GV will answer then I have to press * + my pin and it will grant me access to my Voice mail, you dont have to register any of your GV numbers with Gizmo 5. not the call will not be limited to 3 min.
the only way to do it is the way I have posted. Just use a different set of G5 + GV number .
You don't have to register this set with SS, and the call is free and unlimited because you can call any phone number, and this number just happen to be your own main GV number wish is register with SS. you can use the same SIP phone, Soft phone or PAP2T to dial. Because is going out using a different Set Observe in my dial plan I am using various GV numbers for different purposes , only your main GV number must be register with Sipsorcery.
In this case when you dial 7865551212 (my main GV number), Sipsoucery will use my other set of GV + G5 (not registered with SS, to call my number, once is connected the phone will ring 4 times and GV will answer then I have to press * + my pin and it will grant me access to my Voice mail, you dont have to register any of your GV numbers with Gizmo 5. not the call will not be limited to 3 min.
Code: Select all
when /^7865551212/
sys.Log(" Dialing USA #{req.URI.User} through Google Voice service\t")
Sys.Log("*****************************************************\n")
sys.GoogleVoiceCall("GV3_UserName","GV3_Pass","1747XXXXX30","#{req.URI.User}") I tried this and it works, but you will hear the extra call waiting tones during the 4 rings before your greeting starts. Now, maybe it's better to use the Gizmo5 voicemail which I can access directly from my main GV account registered with SS.......but how do I setup an incoming dial plan to first ring my main GV account for about 10 seconds and if not answered, then forward to my Gizmo5 voicemail...! I'm just thinking it may be better to use the Gizmo5 voicemail instead of the GV voicemail.
Here's what I tried.....but I can't get the incoming dialplan to work. What am I missing?????
if sys.In then sys.Dial("#{sys.SSAccount}@local", 10)
else sys.Dial("1747xxxxxxx@Gizmo")
end
Does something go before this?
Here's what I tried.....but I can't get the incoming dialplan to work. What am I missing?????
if sys.In then sys.Dial("#{sys.SSAccount}@local", 10)
else sys.Dial("1747xxxxxxx@Gizmo")
end
Does something go before this?
Yes just make sure when you add your G5 number to your GV account, you set it as Gizmo not as cell, home or works. you should be able to get your confirmation call.slappydan wrote:Is there some trick to getting your gizmo5 number verified in google voice? I click the verify button in google voice, but the call never comes into my gizmo 5 software
And if you sip phone is register @ Sipsorcery make sure you forward yor G5 number to your SS account, Gizmo 5 will forward any sip call for free.
so you go into the your Gizmo 5 account and forward all calls to your [email protected].
I hope it help
If I don't have the following incoming logic in sipsorcery, then it doesn't work and I get a fast busy when dialing out using sys.GoogleVoiceCall():Josemiami wrote:You should not put anything for your incoming plan, just keep it blank in order to make calls from Google Voice as per SipSorcery, I have not tried any incoming dial plan yet but Aaron suggested to leave blank.
Code: Select all
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.Respond(480, "#{sys.Username} Not available")
else
sys.Dial("Enter Number@Gizmo5",30)
sys.Respond(480, "#{sys.Username} Not available")
end
else
# Do your OUTGOING call processing customisations here.
end
Thanks. I'm thinking it could be a problem with the firewall at work, I'm going to try again when I get home.Josemiami wrote:Yes just make sure when you add your G5 number to your GV account, you set it as Gizmo not as cell, home or works. you should be able to get your confirmation call.slappydan wrote:Is there some trick to getting your gizmo5 number verified in google voice? I click the verify button in google voice, but the call never comes into my gizmo 5 software
And if you sip phone is register @ Sipsorcery make sure you forward yor G5 number to your SS account, Gizmo 5 will forward any sip call for free.
so you go into the your Gizmo 5 account and forward all calls to your [email protected].
I hope it help
synchron wrote:If I don't have the following incoming logic in sipsorcery, then it doesn't work and I get a fast busy when dialing out using sys.GoogleVoiceCall():Josemiami wrote:You should not put anything for your incoming plan, just keep it blank in order to make calls from Google Voice as per SipSorcery, I have not tried any incoming dial plan yet but Aaron suggested to leave blank.
SynchronCode: Select all
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.Respond(480, "#{sys.Username} Not available") else sys.Dial("Enter Number@Gizmo5",30) sys.Respond(480, "#{sys.Username} Not available") end else # Do your OUTGOING call processing customisations here. end
This logic will work without an incoming dial plan:
Code: Select all
####################################################
######## OUTBOUND CALL FROM GOOGLE VOICE ##########
####################################################
sys.Trace = false
sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")
if sys.In
end
else
sys.GoogleVoiceCall("GV_UserName", "GV_Password", "1747XXXXX99", req.URI.User.to_str)
endAs Per AaronAaron
Site Admin
Joined: 12 Jul 2007
Posts: 1643
PostPosted: Thu Aug 20, 2009 11:27 pm Post subject: Reply with quote
Hi David,
Your dialplan looks to be missing a few ends so I'd expect you're getting a syntax error.
You could try changing your dialplan to:
Code:
sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")
case req.URI.User
when /^911$/ then sys.Dial("sipgate")
else sys.GoogleVoiceCall("email","password","my_GIZMO5_#","#{req.URI.User}")
end
Regards,For your incoming calls you don't need a dialplan the default behaviour of sipsorcery will forward the calls to your registered accounts without the need for one. Just make sure the incoming dialplan setting si blank on your SIP account.
Aaron
There is this dial plan without incoming neither all working:
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]","Main_GV_Pass","1747XXXXX99","#{req.URI.User}")
else
#Route Outgoing Via Dialer Length
dialer_length = req.URI.User.Length.to_s
case dialer_length
when /^10/
sys.Log(" If 10 digits, add the 1 and dial provider.\t")
sys.Log("--------------------------------------------------------------------------\n")
sys.GoogleVoiceCall("[email protected]","Main_GV_Pass","1747XXXXX99","1#{req.URI.User}")
when /^11/
sys.Log(" If 11 digits, dial provider.\t")
sys.Log("--------------------------------------------------------------------------\n")
sys.GoogleVoiceCall("[email protected]","Main_GV_Pass","1747XXXXX99","#{req.URI.User}")
end
endOK, I can comment out all the if sys.In logic (like what is stated above) and GV dialing out will work but without the incoming logic, anyone calling my GV number will go to voice mail and my ATA phone won't ring. It needs the logic specifying my registered G5 provider entry in sipsorcery in order to ring the phone, specifically the check for if sys.IsAvailable(). This assumes the G5 account was created with all the defaults, i.e. no forwarding to [email protected], etc.Josemiami wrote:
This logic will work without an incoming dial plan:
Code: Select all
#################################################### ######## OUTBOUND CALL FROM GOOGLE VOICE ########## #################################################### sys.Trace = false sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.") if sys.In end else sys.GoogleVoiceCall("GV_UserName", "GV_Password", "1747XXXXX99", req.URI.User.to_str) end
Synchron
synchron wrote:If I don't have the following incoming logic in sipsorcery, then it doesn't work and I get a fast busy when dialing out using sys.GoogleVoiceCall():Josemiami wrote:You should not put anything for your incoming plan, just keep it blank in order to make calls from Google Voice as per SipSorcery, I have not tried any incoming dial plan yet but Aaron suggested to leave blank.
Code: Select all
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.Respond(480, "#{sys.Username} Not available") else sys.Dial("Enter Number@Gizmo5",30) sys.Respond(480, "#{sys.Username} Not available") end else # Do your OUTGOING call processing customisations here. end
Synchron
synchron, may be this will help you, just do the changes for your dial plan.
jvwelzen wrote:Hi
First I recommend to start testing with only a out dialplan
If your outdialplan is working you can start testing with a in dialplan
make sure you created 2 dialplans one for in and one for out
In your In dialplan you should changetoCode: Select all
if sys.IsAvailable() sys.Dial("magicyes@local&006684123456@smslisto", 35) sys.Dial("sip:[email protected]") else sys.Dial("006684123456@smslisto", 35) sys.Dial("sip:[email protected]") end
Code: Select all
if sys.IsAvailable("magicyes","sipsorcery.com") then sys.Dial("magicyes@local&006684123456@smslisto", 35) sys.Dial("sip:[email protected]", 30) sys.Respond(486, "User Busy") else sys.Dial("006684123456@smslisto", 35) sys.Dial("sip:[email protected]", 30) sys.Respond(480, "Not available") end
That's the ATA I have (kept it when Vonics went oob). Pretty straighforward with entering the same un/pw with what you log on the web at the sipsorcery website. The server and domain under SIP Proxy is just sip.sipsorcery.com and the local signaling is port 5060. This is enough to get you registered and a green Voip light (although your phone won't ring yet and your dial outs wil produce a fast busy).slappydan wrote:Anybody have instructions on setting up a MTA6328-2re with the SIP sorcery info? I've tried, and I can't get it to register.
The challenge with sipsorcery is the dialplan you pick and how you set it up on the Silverlight website. Click the blog button on the main page to get in on all the proper instructions.
Synchron
Thanks, I'll try again.synchron wrote:That's the ATA I have (kept it when Vonics went oob). Pretty straighforward with entering the same un/pw with what you log on the web at the sipsorcery website. The server and domain under SIP Proxy is just sip.sipsorcery.com and the local signaling is port 5060. This is enough to get you registered and a green Voip light (although your phone won't ring yet and your dial outs wil produce a fast busy).slappydan wrote:Anybody have instructions on setting up a MTA6328-2re with the SIP sorcery info? I've tried, and I can't get it to register.
The challenge with sipsorcery is the dialplan you pick and how you set it up on the Silverlight website. Click the blog button on the main page to get in on all the proper instructions.
Synchron
yea I seen that with the briding of the calls in a parking lot. I have not have a few others been able to get it to work.. I did get sip sorcery to work via an ATA but I have to use an FXO card just to use it with trixbox. sip Sorcery should be able to work via trunk configs I just cant seem to get the correct settings. It will register but when calling out I get all circuits are busy
I still can't get my ATA to register, I about ready to give up on itsynchron wrote:That's the ATA I have (kept it when Vonics went oob). Pretty straighforward with entering the same un/pw with what you log on the web at the sipsorcery website. The server and domain under SIP Proxy is just sip.sipsorcery.com and the local signaling is port 5060. This is enough to get you registered and a green Voip light (although your phone won't ring yet and your dial outs wil produce a fast busy).slappydan wrote:Anybody have instructions on setting up a MTA6328-2re with the SIP sorcery info? I've tried, and I can't get it to register.
The challenge with sipsorcery is the dialplan you pick and how you set it up on the Silverlight website. Click the blog button on the main page to get in on all the proper instructions.
Synchron
slappydan - not sure I can help you much more. Were you able to ever get the same ATA to work with mj and mjproxy? Or just with the regular SIP credentials prior to 06/09?
BTW, looks like after a couple of short outages this week with the sipsorcery server and website, Aaron has ceased allowing new accounts and to check back in a couple of weeks. Since Silverlight is the main engine running the sipserver and the website, I'm not surprised SS went into overload now that there have been so many new accounts since GV started going into full force recently. I'm convinced many of those accounts came from right here at mjsupport.
Synchron
BTW, looks like after a couple of short outages this week with the sipsorcery server and website, Aaron has ceased allowing new accounts and to check back in a couple of weeks. Since Silverlight is the main engine running the sipserver and the website, I'm not surprised SS went into overload now that there have been so many new accounts since GV started going into full force recently. I'm convinced many of those accounts came from right here at mjsupport.
Synchron
I never tried with mjproxy, but I did have it working prior to 6/9. I'll keep playing with it, but I'm pretty much resigned to using the dongle.synchron wrote:slappydan - not sure I can help you much more. Were you able to ever get the same ATA to work with mj and mjproxy? Or just with the regular SIP credentials prior to 06/09?
BTW, looks like after a couple of short outages this week with the sipsorcery server and website, Aaron has ceased allowing new accounts and to check back in a couple of weeks. Since Silverlight is the main engine running the sipserver and the website, I'm not surprised SS went into overload now that there have been so many new accounts since GV started going into full force recently. I'm convinced many of those accounts came from right here at mjsupport.
Synchron