Sipsorcery down?

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degarb
Posts: 40
Joined: Tue Sep 01, 2009 9:18 pm

Post by degarb » Fri Nov 13, 2009 6:22 pm

ilanesh wrote:hongkongpom, sorry for my english that I confused you, it is not my native language.
As to dyndns.org, it is a free dns server, no need to install, this one I use for my router cause ,my IP from time to time changes.
For to forward to my IP strait to the ATA I use cjb.net, it is another free dns server. Both no need to install.
When you add your sip providers on sipsorcery then it is automatic on: your username@sipsorcery.com
But nastead you can do this way: If your provider is for example gizmo5 then you use when for example your username is: 12345 for gizmo5 then this way:
12345@your dyndns or cjb.net username.dyndns.org:5060 or cjb.net:5060, when your ATA is on port 5060. I wish I could better explain it with my easy english, but this way it go`s strait to your ATA you will have also better sound.

You can sign up for dyndns.org and then use any provider with this forwarding this way to your ATA. Also enable on your router your ATA`s IP and your ATA`s Port, the default is 5060. I gave my pap2 a stable internal IP and put this IP in my router.

The add for dynds is: www.dyndns.org
and the other free is: www.cjb.net
I found out that the cjb.net is more stable or less problems when forwarding with voip.
Good luck. :-)
Firstly, I have limited time and tired eyes, so forgive me daftness too. If I am reading correctly, you have a way to set a sipsocery setting to allow incoming calls if their server goes down. (I dont under stand how a setting on the sipsorcery end will allow any change if they drop off the face of the earth.)

Anyway, on faith, I tried sipsorcerylogin>sip providers>double click sipgate on first part> Register contact=sipgateusername@dynamic ip:5060 then update And tried sip:sipgateusername@dynamic ip:5060 Then I turned on nat mapping.

Unfortunately this alone broke the phone service. So, something is wrong.

ilanesh
Posts: 72
Joined: Sun Oct 19, 2008 12:18 pm

Post by ilanesh » Fri Nov 13, 2009 7:28 pm

degarb, you forgot something.

sipgateusername@dyndnsusername.dyndns.org:5060

degarb
Posts: 40
Joined: Tue Sep 01, 2009 9:18 pm

Post by degarb » Fri Nov 13, 2009 8:46 pm

ilanesh wrote:degarb, you forgot something.

sipgateusername@dyndnsusername.dyndns.org:5060
But I changed
sip:degarb@sipsorcery.com to 8008367e0@akronedge.info:5060

akronedge.info is my ip and website. so I don't need dyndns.org

I hope I am forgetting something, but what?

Are you using a pap2t? Perhaps, setting for outgoing proxy could be different than yours. In theory, each line should have different settings on any ata (or flexibility) and incoming and outgoing use differing services. I wrestled with settings for 5 hours to get ones that work, bewildered with those that I don't understand fully (like most). But It seems using an outgoing proxy never worked on the pap2t, but don't remember fully.

mxnerd
Posts: 63
Joined: Fri Jul 17, 2009 1:50 am

Post by mxnerd » Fri Nov 13, 2009 10:35 pm

I guess probably ilanesh has a local version?

Or probably sipsorcery.com is back and he thinks his method worked?

I really don't see how it will work if Sipsorcery.com is down or there is no local version running.

ilanesh
Posts: 72
Joined: Sun Oct 19, 2008 12:18 pm

Post by ilanesh » Fri Nov 13, 2009 11:18 pm

degarb, when you use 8008367e0@akronedge.info:5060 so you for sure should have a username for your akronedge.info, then you need to put your username befor akronedge.info like
8008367e0@akronedgeusername.akronedge.info, but I am not sure if this works with this one, I use always dns and this is then my ip adress.
I used it all the time when sipsorcery went down and got always incoming calls from my friend, but I also leave it always forwardet this way also when sipsorcery is not down, I do not any changes.
Yes, I use pap2-na, but no need to register pap2 with the free dns, I did it only on sipsorcery so it is going strait to my ip this way thats all. But if you don`t like you do not need it or when it makes to you problems, I just like the Idea that when all gets wrong then it keeps on ringing when someone calls me and we can still talk. And also no sound problems this way, the sound is way better then when it has to go through the sip provider.
No I have no local sipsorcery, I use the public one I even could not use it cause I have no server for to leave day and night the machine running.
Hope you can get it to work. Maybe it works only with a dns.

ilanesh
Posts: 72
Joined: Sun Oct 19, 2008 12:18 pm

Post by ilanesh » Fri Nov 13, 2009 11:25 pm

sorry I only saw it now. You had degarb@sipsorcery.com and you should put it this way then.
degarb@8008367e0.akronedge.info:5060
if is 8008367e0 your akronedge username, hope this helps now.
Ila

snvv
Posts: 153
Joined: Sun Oct 26, 2008 10:43 pm

Post by snvv » Fri Nov 13, 2009 11:44 pm

Hello,

I tried myproviderID@dnsID.dyddns.org:5060
I changed Ans Call Without Reg: yes

nothing happens.

Then I changed dnsID.dyddns.org to my actuall IP but nothing happens too.

The tests made with SPA3102 & PAP2.

...........

During the time SS was down I used voxalot.
Surprisingly, my ATA staid on line all days without any disconnect. On the other hand using SS once (at least) per day my ATA looses connection with my router and I have to restart the router to get connection .

Regards,
snvv

hongkongpom
Posts: 29
Joined: Wed Oct 14, 2009 10:04 am

Post by hongkongpom » Sat Nov 14, 2009 2:42 am

ilanesh wrote:hongkongpom, no you do not need it in your router, it is just when you would like to use it. I works without to put dyndns.org into your router, it is very simple all, I could walk you through, you can pm me and I give you my gizmo5 number, but you still need sipsorcery, it is just a good way that when sipsorrcery is down then you still can receive your incoming calls without sipsorcery, but only when you on your ATA put "receive incoming calls without register", otherwise it will not work, and outgoing calls I did via sipbroker when sipsorcery was down. And you do not need to tuch your dialplan at all, leave your dialplan as it is, it is not affecting your dialplan.
Hi ilanesh,

I am still not clear about how this will work so let me see if my understanding is correct.

Please write the steps for us on this forum because I think many people would like to do this.

You say I can forward the incoming VoIP calls that arrive at SS, directly to my SPA3102. I do this on the SS screen. I just change my SIP forwarding URI to my ddns username:5060

My public IP address given to me by my ISP is dynamic and so I must use a ddns service which updates my IP address automatically as it is changed.

So, to connect my SPA3102 I can use the username that I have signed up for with the ddns company. Even if my IP address changes, it doesn't matter because the ddns company knows.

In order for the ddns company to know my IP address I must have some sort of client running on my end. Some routers have this function. If not then I have to download & run a ddns client on a PC connected in my network.

You say I don't need to install anything on my end but how does the ddns server know that my IP address has changed?

I can see if I could forward the calls directly to my ddns username:5060 that would be great. The calls would then be forwarded to my SPA.

If SS is offline though then the forwarding will not happen. You say when SS is online, you can forward the calls, but if it is online then why do you need to forward the calls? The next time the SS server goes offine the forwarding will stop.

Please correct my misunderstanding because I would love to use this method. :)

mxnerd
Posts: 63
Joined: Fri Jul 17, 2009 1:50 am

Post by mxnerd » Sat Nov 14, 2009 3:21 am

There is no way this will work without a DDNS client, either on Windows/Linux/Mac machine or on the router. No-IP.com or Dyndns.com, etc provides free DDNS service.

Many routers released in past few years already have DDNS built-in, and you can enter your DDNS account in the router.

So DDNS client must be running somewhere, either your PC or router, unless you got a static IP.

hongkongpom
Posts: 29
Joined: Wed Oct 14, 2009 10:04 am

Post by hongkongpom » Sat Nov 14, 2009 3:29 am

ilanesh wrote:And also no sound problems this way, the sound is way better then when it has to go through the sip provider.
I think SS only does the SIP UDP signalling ie setting up the call between two parties. I don't think it handles the RTP traffic so it won't have any effect on the call sound quality.

This is from the MSS FAQ which also applies to SS

How does My SIP Switch interact with audio codec?
No! My SIP Switch doesn't handle media at all, so there is not even a remote possibility that it affects audio quality.

http://www.mysipswitch.com/faq.aspx#12

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