Hello,
I like the idea of mysipswitch.com very much even if I had some difficilties with voipcheap before, but that seems not to be mysipswitch's fault.
So I have a question: How long is this system already in use? I heard only for some months so far?
How will the future be like? Will it last like a normal VoIP provider or is it just a short termed project?
I would even be willing to pay some bucks per year if I knew I can rely on it.
Technical question: Is the whole calling traffic going through your server or only the initial part?
Is this the only online version of a "SIP Proxy" available?
Anyway: Keep on with your good work!
Future of mysipswitch.com
Re: Future of mysipswitch.com
I first started playing with Mysipswitch in November 2006 and have been using it more or less constantly since early 2007. Although Aaron often reminds users that there are no guarantees about server availability etc I have found the service excellent.thwill wrote:So I have a question: How long is this system already in use? I heard only for some months so far?
The project is open source - meaning anyone can develop/add tt & correct code. The guys that have been developing it up to this work for http://www.blueface.ie, arguably the best VOIP provider in Ireland. Many mysipswitch users (including me) are also Blueface customers so I guess we are in a way supporting the project by keeping Aaron and Guillaume in work!thwill wrote: How will the future be like? Will it last like a normal VoIP provider or is it just a short termed project?
My understanding is that the SIP switch does NOT handle the traffic (as in speech traffic) it is only involved in the signalling / call setup etc prior to the audio packets being transmitted. All audio packets are point to point between your ATA/softphone and your VOIP gateway or another VOIP user. Mysipswitch performs two main functions allowing the user much greater control over how incoming and outgoing calls are routed:-thwill wrote: Technical question: Is the whole calling traffic going through your server
1) SIP Registration
This allows the sipswitch to register with many VOIP providers on your behalf allowing you to receive (or forward) incoming calls from each provider to your ATA or another ip address.
2) Intrepreting dial plans for outgoing call and seting up call with various VOIP providers.
This allows you to place outgoing calls via many different providers without having to reconfigure your ATA or softphone each time.
Hope this helps!
Thanks
Martin, thank your for your long and accurate answer.
I have registered myself with mysipwitch already (two accounts) and know how to use it, especially the dialing plan is cool.
I want to set up one account for my family in the Philippines and so I can always configure everything from here in Germany.
The second account is for myself.
So especially point number 2) of your explanation is perfect for me!
I have registered myself with mysipwitch already (two accounts) and know how to use it, especially the dialing plan is cool.
I want to set up one account for my family in the Philippines and so I can always configure everything from here in Germany.
The second account is for myself.
So especially point number 2) of your explanation is perfect for me!
Let me join the group is saying how excellent this service is. I always had a problem registering more than one SIP account on my ATA. The ATAs in general and until recent are programmed to register 1 or 2 (at the most) SIP accounts. A friend at Voxilla told me about the link. I have been using it for less than a month. It solved all the problems I was facing in a blink of an eye. The service is so far reliable. The only problem I am facing is my inability to program a complex dialplan for outgoing calls. However, I am satisfied with what available.
As I read earlier, this project is for test and may not continue. This makes me worried because I honestly rely on it 100%. I know people don’t like to commit themselves to promises they cannot make. Nevertheless, it is important to know more information on how long this project will continue. I don’t want to wake up one day and find myself disconnected. More information in this regard will be highly appreciated.
I also want to add that strengthening the outgoing dialplans will also be appreciated.
Keep the good work folks.
As I read earlier, this project is for test and may not continue. This makes me worried because I honestly rely on it 100%. I know people don’t like to commit themselves to promises they cannot make. Nevertheless, it is important to know more information on how long this project will continue. I don’t want to wake up one day and find myself disconnected. More information in this regard will be highly appreciated.
I also want to add that strengthening the outgoing dialplans will also be appreciated.
Keep the good work folks.
Hi yf,yf wrote:The only problem I am facing is my inability to program a complex dialplan for outgoing calls. However, I am satisfied with what available.
The funny thing is when we first put up the IP switch there was an "Advanced Dial Plan" option where instead of using the current format the dial plan could be configured as a C# method with all the programming constructs and flexibility you could ever possibly want. As an example:
Code: Select all
public class SIPSwitchAdvConfig : SIPProxyRuntimeConfig
{
public override SwitchCallStruct SwitchCall(IPEndPoint localEndPoint, IPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
// ===== Cut below here and paste into the advanced configuration textbox.
SwitchCallStruct switchCallStruct = SwitchCallStruct.Empty;
string dst = sipRequest.URI.User;
switch(dst)
{
case "100":
// Anonymous calls to Blue Face.
switchCallStruct = new SwitchCallStruct("anon", null, "sip:303@sip.blueface.ie", "<sip:anon@sip.mysipswitch.com>", "<sip:303@sip.blueface.ie>", "sip.blueface.ie");
break;
case "101":
// Anonymous call to FreeWorldDialUp.
switchCallStruct = new SwitchCallStruct("anon", null, "sip:612@fwd.pulver.com", "<sip:anon@sip.mysipswitch.com>", "<sip:303@fwd.pulver.com>", "fwd.pulver.com");
break;
default:
break;
}
if(Regex.Match(dst, "^102$").Success)
{
// To place an authenticated call the username and password must be valid and in most cases the from host must match the provider's domain.
// If the call belows gets a nocredit message then it has been successfully authenticated.
switchCallStruct = new SwitchCallStruct("sipswitch", "password", "sip:012222222@sip.blueface.ie", "<sip:sipswitch@sip.blueface.ie>", "<sip:012222222@sip.blueface.ie>", "sip.blueface.ie");
}
return switchCallStruct;
// ===== Cut above here and paste into the advanced configuration textbox.
}
}
Regards,
Aaron