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MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Post by MikeTelis » Sat Nov 14, 2009 5:09 am

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.
I'll second that. Besides, forwarding incoming calls directly to your ATA you'll lose all advantages provided by "In" dialplan plus (should I say 'minus'??) sys.GoogleVoiceCall won't work because SS won't receive callbacks from GV.

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

Post by ilanesh » Sat Nov 14, 2009 10:50 am


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

Post by ilanesh » Sat Nov 14, 2009 12:00 pm

Provider Name gizmo5
Username 1747XXXXXXXXX
Password XXXXXXX
Server sip:proxy01.sipphone.com
Register Yes
Register Contact sip:mygizmo5username@myddnsusername.cjb.net:5060
Then click update,

This is on your SS all.
Then check if it is registered on SS, when registered all should work.

On your ATA or pap2-na
if you have ss on line 1 then enable line 1
Make call without register=yes
answer call without register=yes
Nat Mapping enable=yes
Nat keep alive enable=yes
enable IP dialing=yes
Thats it.

You can also use the username from another sip provider anstead your gizmo5 username and forward the other sipprovider the same way like I did it here with whatever ddns you are using, but must sign up and create an account and from time to time update it and they have always your ip and when it changes they update it.
Good luck.

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

Post by degarb » Sat Nov 14, 2009 6:27 pm

I think I get it now. Maybe. I am using siggate not gizmo5

I do have a gizmo5 account, however, I got the dialing plan of sipgate. If you don't give me the dialing plan, or point me to thread, I might be able to pull this off.

I tried putting in your posted settings equivalent with sipgate and my gizmo5, but no avail. I think because my dialing plan is sipgate. Then again, I don't know.

I put dial and answer without registering to yes. It doesn't seem to change any results with old or new settings. And ipdialing to yes seems no affect. But turning on nat mapping seems to kill incoming audio on the phone.

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

Post by ilanesh » Sat Nov 14, 2009 7:29 pm

degarb, this has nothing to do with your dialing plan, you dialing plan leave as it is, the only thing is importent that it is registered, but there may be a problem that your sipprovider refuses incoming and outgoing calls without register, or maybe it is just your router.
I have a linksys router and have everything forwardet to the ATA ports. I am sorry when it is not working with you. But honestly with dyndns I had also such problems like you so this is why I use now cjb.net. You can also try no-ip, but I really do not understand what you mean with dialing plan. Shall I try to give you a call? You can pm me if you wish.
So I give you also my gizmo5 number, I try to help you if it is possible, but also every router acts differently.
It took me many months to get my voip working like it is working now, believe me with many tears till it worked finally.
Also for each person works different settings. What I am actually doing is that people just call my IP directly but then your voicemail will not work this is the disadvantage.

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

Post by hongkongpom » Sun Nov 15, 2009 2:12 pm

ilanesh wrote:Provider Name gizmo5
Username 1747XXXXXXXXX
Password XXXXXXX
Server sip:proxy01.sipphone.com
Register Yes
Register Contact sip:mygizmo5username@myddnsusername.cjb.net:5060
Then click update,

This is on your SS all.
Then check if it is registered on SS, when registered all should work.

On your ATA or pap2-na
if you have ss on line 1 then enable line 1
Make call without register=yes
answer call without register=yes
Nat Mapping enable=yes
Nat keep alive enable=yes
enable IP dialing=yes
Thats it.

You can also use the username from another sip provider anstead your gizmo5 username and forward the other sipprovider the same way like I did it here with whatever ddns you are using, but must sign up and create an account and from time to time update it and they have always your ip and when it changes they update it.
Good luck.
Hi ilanesh,

How is the ddns service that you setup as the Register Contact getting the ATA public IP address?

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

Post by ilanesh » Sun Nov 15, 2009 2:33 pm

Hi hongkongpom, I am very pleased with cjb.net, till now I have had no problem with it. I is your external IP what they get, and then it is forwardet to your external ip and it will ring your ATA or pap2-na`s phone who is connected, but you must have sipsorcery logged in in your pap2, and your voip provider must be logged in in on your sipsorcery, then go to www.cjb.net and sigh up there and you choose the username you want and the password you want and so you will get your cjb.net adress who is then your external ip adress. But you have to go from time to time to their website and update your ip, you just put your username and password in and click on udate my ip and thats it. They have also software for to autamted updating, you will find all on their site.

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

Post by hongkongpom » Sun Nov 15, 2009 5:02 pm

ilanesh wrote:Hi hongkongpom, I am very pleased with cjb.net, till now I have had no problem with it. I is your external IP what they get, and then it is forwardet to your external ip and it will ring your ATA or pap2-na`s phone who is connected, but you must have sipsorcery logged in in your pap2, and your voip provider must be logged in in on your sipsorcery, then go to www.cjb.net and sigh up there and you choose the username you want and the password you want and so you will get your cjb.net adress who is then your external ip adress. But you have to go from time to time to their website and update your ip, you just put your username and password in and click on udate my ip and thats it. They have also software for to autamted updating, you will find all on their site.
Thanks ilanesh,

This would not work for me because my ISP changes my IP address every 3 hours.
My ATA cannot do ddns so unfortunately I am stuck.

The other problem with adding your ddns SIP URI to your SS account is if SS goes offline then the forwarding must also not work?

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

Post by ilanesh » Sun Nov 15, 2009 5:18 pm

it works for incoming calls, not for outgoing, I used x-lite the 3 days it was down when calling pstn phone, but voip2voip it worked cause I use on my pap2-na the sipbroker gateway. So I had no problems, the DNS is specialy for the changing dynamic ip, this is why I use it, when you have a static one you do not need dns at all.

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

Post by hongkongpom » Mon Nov 16, 2009 1:31 am

ilanesh wrote:it works for incoming calls, not for outgoing, I used x-lite the 3 days it was down when calling pstn phone, but voip2voip it worked cause I use on my pap2-na the sipbroker gateway. So I had no problems, the DNS is specialy for the changing dynamic ip, this is why I use it, when you have a static one you do not need dns at all.
Ok I think what you are saying is you set up the forwarding on your incoming VSP so it forwards directly to your ATA. It will forward to either your static IP address:3060 or your ddns username:3060

SS is not in the process, it is bypassed.

If static IP then yes this works well. If not then I must use ddns. But to use ddns I must install the ddns client in my home. My ATA cannot do ddns. My router cannot to ddns so my only option is to install the ddns on my PC.

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