How to Use MySipSwitch

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emoci
Posts: 127
Joined: Mon Aug 20, 2007 11:27 pm

How to Use MySipSwitch

Post by emoci » Wed Oct 24, 2007 12:52 am

There has been a few new features lately, so here's a quick summary of what you can do with MySipSwitch right now:


1.Register your SoftPhone or ATA with MySipSwitch
User: SSUser
Password:SSPassword
Host: sip.mysipswitch.com

WRTP54G-ER (thanks to Fixup)
Image

Other Devices (via the MySipSwitch Blog)
http://www.mysipswitch.com/wordpress/in ... iguration/

If you'll be using MySipSwitch for Dial Plan configuration, all you need in your ATA dial plan line is ([*x][*x].) (this applies only to Linksys devices)

2. Configure Your Providers

Image


Provider Name: Any descriptive name will do (keep it one word and simple)
Username: The corresponding provider username
Password: Your Password
Server: Your third party server (eg. us.voxalot.com)

*If all you want to do is make outgoing calls with this provider stop here*

Show Provider Advanced Call Setting: Use this checkbox if you require any of the following settings
Outbound Proxy: A few providers require that you specify this
From Header: If your provider supports adjusting Caller ID, this is the place to do it.

To send through a specific CID (can include name as well):
<sip:NumberToSend@sip.provider.com>
or
"MyName" <sip:NumberToSend@sip.provider.com>

To pass through the CID of the incoming caller (preferrable if you are using this provider for Call Forwarding):
${fromname} <sip:${fromuriuser}@sip.mysipswitch.com>

Please Note: A number of providers use the From Header value for authentication purposes, and may not work well when making the alterations above. As a result not all providers will be compatible with the From Header changes from above.

Custom Header: Here you can enter a number of custom strings. Once more keep in mind that this may interfere with provider functioning, and may not work with all providers.

Some of the options:
-P-Src-IP: 127.0.0.1 (replace 127.0.0.1 with your IP). This was put in place to deal with FUP issues with BetaMax providers, results on effectivity have been mixed
- 127.0.0.1 (replace 127.0.0.1 with your IP). You may also want to try entering your IP without prefacing it with P-Src-Ip and see if this more effective for your purposes in regards to the BetaMax FUP issue.

*Stop here if you do not want to have calls to this provider ring at MySipSwitch*

Register with this provider: Check this box if you require registration for inbound calls to ring at MySipSwitch
Registration Contact: This where you want calls placed to this provider to ring.

Having calls ring at MySipSwitch: Set the contact address as sip:YourUsername@sip.mysipswitch.com

Having calls ring at another destination: eg. sip:123456@us.voxalot.com (would allow you to have calls for this provider ring at VoXalot for example)

If you want to specify a registration port (not always neccessary) you could enter sip:123456@us.voxalot.com:5060 (where 5060 is the port)

Registration Expiry: By default registrations are renewed every 3600 sec (1 hour), change this value as needed

All of the following fields are optional and should only be adjusted if your provider has specific requirements:
Register Server: Optional
Auth Username: Some providers use a different username to authenticate incoming calls (as opposed to outgoing calls). If your provider requires this (sometimes known as Auth ID) enter it here.
Domain/Realm: Some services require that you specify both a Proxy and a Domain for registration. Here is where you would enter the Register Domain if this is required...

Don't forget to save. Now this provider will appear on your list. You can edit it if neccessary. If you chose to register the provider, use the Monitoring Tab to check the registration status after saving.

Specific Provider Configurations:
http://www.mysipswitch.com/forum/viewtopic.php?t=602


3. Configure your dial plans (Within MySipSwitch)

MySipSwitch offers 3 possible methods to write your dialplans (pick one and stick to it :wink: ):

-The New Dial Plan syntax (more info coming soon)
How to compose a transparent Dial Plan

-The Old Dial Plan syntax
The old dialplan syntax can still be used. However unlike the New Syntax, or Ruby, it does not make use of the pre-configured providers as described above, but requires that all provider information be entered as part of the Dial Plan

Advanced Ruby Dial Plans (require some basic programming, but introduce a multitude of advanced features)

Default Ruby Dial Plan Demystified
Inbound Call Managment with Ruby
Available Ruby Modules
More Available Ruby Modules
Ruby Example with explanation
More Ruby Use Cases



For the Techies:

Source Code: http://www.mysipswitch.com/forum/viewtopic.php?t=156
Also see:
http://www.mysipswitch.com/forum/viewtopic.php?t=277
http://www.mysipswitch.com/forum/viewtopic.php?t=166
http://www.mysipswitch.com/forum/viewtopic.php?t=100
Executable APP (Windows/Linux):
http://www.mysipswitch.com/forum/viewtopic.php?t=261
Also see: http://www.mysipswitch.com/forum/viewtopic.php?t=317
TelNet Troubleshooting:
http://www.mysipswitch.com/forum/viewtopic.php?t=315
MySipSwitch on The Amazon Cloud:
http://www.mysipswitch.com/forum/viewtopic.php?t=265
Sip Signalling Test Tool:
http://www.mysipswitch.com/forum/viewtopic.php?t=114
Wiki/Voip-Info: http://www.mysipswitch.com/forum/viewtopic.php?t=162

I'll update these instructions as I test, Aaron and team if you feel something could be said better in another way, or needs to be edited feel free to do so

Upcoming Updates:
-Quick Guide on the New Dial Plan Syntax
-A WalkThrough for Ruby Dial Plans
Last edited by emoci on Thu Oct 23, 2008 2:57 am, edited 23 times in total.

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Aaron
Site Admin
Posts: 4563
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Wed Oct 24, 2007 1:27 am

Thank you very much emoci, thats a great guide. Guillaume and I have had a few attempts at a user guide but we get wrapped up in technical explanations or obscure features; your's is much clearer.

If you don't have any objections we'll incorporate your guide into the main help pages?

Regards,

Aaron

emoci
Posts: 127
Joined: Mon Aug 20, 2007 11:27 pm

Post by emoci » Wed Oct 24, 2007 2:31 am

Aaron wrote:Thank you very much emoci, thats a great guide. Guillaume and I have had a few attempts at a user guide but we get wrapped up in technical explanations or obscure features; your's is much clearer.

If you don't have any objections we'll incorporate your guide into the main help pages?

Regards,

Aaron
Fine by me.... glad to help, it's kinda neat when you see an overview of the things it can achieve!!!

User avatar
TheFug
Posts: 914
Joined: Sat Oct 06, 2007 8:23 am
Location: The Netherlands, North-Holland

Post by TheFug » Wed Oct 24, 2007 5:09 pm

If i understand correctly, the outgoing provider is a provider where this feature is an extra paid sevice or is it the normal service, just like calling with an ATA or softphone, and this happens at the normal "prepay" rates or free days, it's getting a bit complicated for me :) so, lets say i expect a sip call, but i'm at work, not at home, i can route the sip call to my phone number at work, via voipbuster this way:

exten = SipSwitchUsername,1,SwitchCall(vb-user,vb-pass,003120xxxxxxx@sip.voipbuster.com)

or i can add a rule which point to my SipSwitchUsername...?

I guess anyone who calls, calls for free at my (voipbuster) account so i'll pay the expences.
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

emoci
Posts: 127
Joined: Mon Aug 20, 2007 11:27 pm

Post by emoci » Wed Oct 24, 2007 6:03 pm

Depends on where the person will call:

If I call you at SipSwitchUser@sip.mysipswitch.com then:

-You need to have the box checked
-You need to have:
exten = SipSwitchUsername,1,SwitchCall(vb-user,vb-pass,003120xxxxxxx@sip.voipbuster.com)
in dial plans

Your number 003120xxxxxxx will ring. The call will be handled by VoIPBuster and if you have free minutes and this (003120xxxxxxx) is one of the numbers that can be reached for free, than it will be a free call

Now let's say you have a DID Number from another company (eg. A did from CallCentric)

-You perform a registration on the DID, and point to SipSwitchUser@sip.mysipswitch.com
-You need to have the box checked
-You need to have:
exten = SipSwitchUsername,1,SwitchCall(vb-user,vb-pass,003120xxxxxxx@sip.voipbuster.com)
in dial plans

Now if I call your CallCentric DID your phone will ring and the call will be handled by VoipBuster just as before.....

Whenever a call comes in, VoipBuster has to place a call to your number but unless you are calling (having the call sent to) a Non-Free number.....this should fall within your free weekly minutes at VoipBuster

Does that clarify things....sort of?
TheFug wrote:If i understand correctly, the outgoing provider is a provider where this feature is an extra paid sevice or is it the normal service, just like calling with an ATA or softphone, and this happens at the normal "prepay" rates or free days, it's getting a bit complicated for me :) so, lets say i expect a sip call, but i'm at work, not at home, i can route the sip call to my phone number at work, via voipbuster this way:

exten = SipSwitchUsername,1,SwitchCall(vb-user,vb-pass,003120xxxxxxx@sip.voipbuster.com)

or i can add a rule which point to my SipSwitchUsername...?

I guess anyone who calls, calls for free at my (voipbuster) account so i'll pay the expences.

User avatar
TheFug
Posts: 914
Joined: Sat Oct 06, 2007 8:23 am
Location: The Netherlands, North-Holland

Post by TheFug » Wed Oct 24, 2007 11:17 pm

Yes, it slowly does, it's all new to me, also ordered/received some books on the subject, to read, when not online...
there are some options possilble ! i look for options to check, when not at home, so, look how to combine GSM, WiFi, or (free) web based services, to check incomming calls.
MySIPSwitch is a great help in this too.
Thanks for your clear explanations !
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

emoci
Posts: 127
Joined: Mon Aug 20, 2007 11:27 pm

Post by emoci » Thu Oct 25, 2007 1:48 am

TheFug wrote:Yes, it slowly does, it's all new to me, also ordered/received some books on the subject, to read, when not online...
there are some options possilble ! i look for options to check, when not at home, so, look how to combine GSM, WiFi, or (free) web based services, to check incomming calls.
MySIPSwitch is a great help in this too.
Thanks for your clear explanations !
My personal opinion, don't waste money on books, you can learn all you need to know with a few google searches and trying things hands on....

Take a look at this http://www.redflagdeals.com/forums/show ... p?t=454642

What sort of a line do you have at home (that you want to check incoming calls)? If it is a VoIP line, this should be straight forward..... if you can describe your current services a bit, and a general idea of what you want to do, I may be able to make a few suggestions...

User avatar
TheFug
Posts: 914
Joined: Sat Oct 06, 2007 8:23 am
Location: The Netherlands, North-Holland

Post by TheFug » Thu Oct 25, 2007 4:16 pm

emoci wrote:
TheFug wrote:Yes, it slowly does, it's all new to me, also ordered/received some books on the subject, to read, when not online...
there are some options possilble ! i look for options to check, when not at home, so, look how to combine GSM, WiFi, or (free) web based services, to check incomming calls.
MySIPSwitch is a great help in this too.
Thanks for your clear explanations !
My personal opinion, don't waste money on books, you can learn all you need to know with a few google searches and trying things hands on....

Take a look at this http://www.redflagdeals.com/forums/show ... p?t=454642

What sort of a line do you have at home (that you want to check incoming calls)? If it is a VoIP line, this should be straight forward..... if you can describe your current services a bit, and a general idea of what you want to do, I may be able to make a few suggestions...
You are correct, but i needed some general information, and something to read "on the go" (i commute by bus) and to being able asking the right questions :) nice "add up" link ! i've bookmarke that one.
Well, my sittuation here: regular ADSL 1,5Mbps, one modem/router, SPA3102 on LAN port, PC on other LAN port, laptop via WiFi (WPA) POTS is also connected to SPA3102, i phone by a DECT set (3 phones) with CID and adjustable ringtones.
my dialplan at MySIPSwitch:
exten => _00X.,1,Switch(Username,Password,${EXTEN}@sip.voipbuster.com, <sip:0031MyCIDnumber@sip.voipbuster.com>)
exten => _06X.,1,Switch(Username,Password,+316${EXTEN:2}@sip.VoipCheap.com, <sip:0031MyCIDnumber@sip.voipcheap.com>)
exten => _0ZX.,1,Switch(Username,Password,+31${EXTEN:1}@sip.voipbuster.com, <sip:0031MyCIDnumber@sip.voipbuster.com>)

Want to make as much use of free services, for my "needs", if possible, manage/check remotely my VoIP and landline calls....
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

emoci
Posts: 127
Joined: Mon Aug 20, 2007 11:27 pm

Post by emoci » Wed Dec 12, 2007 1:40 am

Made some changes to the original post, especially in regards to optional features of the Dial Plan expression...

If Aaron or gbonnet has some time, do read over it to make sure I didn't mess anything up

satphoneguy
Posts: 131
Joined: Tue Oct 23, 2007 12:54 pm

Post by satphoneguy » Wed Dec 12, 2007 1:52 am

emoci,

great updated overview. this makes it really easy to learn.

thanks

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