Posted: Thu Jan 13, 2011 20:56 Post subject: Does anyone know whats happened to milkfish.org ??
milkfish.org sees to be offline.
They seem to be down and have been since the start of the......decade :?
Is there a mirror out there ? I am specifically interested in some custom configs where I have three or four handsets behind NAT register to their own SIP accounts on an Asterisk box in front of NAT, but also do RTP media hand off from the Asterisk server to the ITSP while keeping signalling through asterisk (REINVITE)
I can probably hack the cfg file myself, but it had some nice info that was clearly laid out for setting up a custom ser.cfg file
Any heads up would be appreciated :)
Cheers
Chris
ps. I am posting here as i will be testing it on a couple of different chipset routers
same problem here. I am a novice in milkfish, just got a new router and wanted to see how I can set it up. did the project die completely? is there a place with a documentation?
If you go to this page and click on the SER link it takes you to http://www.iptel.org/ser/. I registered an account but not sure if the account is a viable substitute for the milkfish username / pw. Specifically, the dd-wrt help describes the field:
Quote:
Milkfish Username:
Here goes your Milkfish Community Forum Username - Registration allocates you yourname.homesip.net
Can the Username / PW fields be configured to work with iptel.org account? This would be useful, now that milfish.org accounts are no longer accepting new accounts.
Can anyone confirm that by changing the "From-Domain" field (in the DD-WRT Milkfish Router config page) to iptel.org configures the Username / PW to work with the iptel account? Is there a way to test this?
Joined: 04 Feb 2007 Posts: 426 Location: Fountain of Youth
Posted: Wed Mar 02, 2011 0:31 Post subject:
It is is an ITSP (I think): it issues you a phone number, however, it only receives calls. Given that it is free, one can not really complain.
Quote:
IPKall is a PSTN to Existing VoIP service ONLY; devices and/or software are not to be logged into IPKall servers. IPKall reserves the right to refuse or terminate service for any reason.
Quote:
Calls will come from 66.54.140.46 (voiper.ipkall.com) or 66.54.140.47 (voiper2.ipkall.com).
I am not sure what the "SIP Phone Number:" should be. I think that the SIP proxy should be my domain that resolves to my WAN IP. Hopefully someone with experience can provide guidance.
SIP (Session Initiation Protocol) is a plain text protocol, similar to SMTP, in that it is designed to manage the signalling and communications between two SIP devices. http://en.wikipedia.org/wiki/Session_Initiation_Protocol
Like an SMTP email header, you have a source address, destiantion address, and a number of commands, such as INVITE, OK, NOTIFY etc allowing you to exchange information about what you are trying to send and how its going to get there.
The media or payload is not managed by SIP, the same way that you have a MIME attachment in an email with the actual content of what you are trying to send. With SIP, thats handled by RTP in the case of voice and video
You register your SIP client, such as XLite or a Polycom IP phone to a SIP regsitration server. In this case it will be the voiper(2).ipkall.com
You give it your username and password. It says
"You're cool. I now have your IP address to send my SIP stuff to you"
Then someone calls your IPKall landline number. IPKall's server then send the call to your SIP client as a SIP call with all of the SIP information. Your SIP client, responds accordingly and rings, answers and sets up the RTP media path between its self and the IPKAll server.
This is similar to sending you an email, with a PDF attached.
SIP doesn't carry the audio.
If you just want to recieve calls on a single handset or Softphone client, then you dont need Milkfish.
I am an absolute beginner at Milkfish, but I have a good understanding of SIP and have been learning about Kamailio (formally OpenSER, which is what Milkfish is built from), SER and OpenSIPS
Where Milkfish is good, is it allows you to have two or more client connections behind the router on the one SIP account. You can also make SIP calls directly between the clients.
Now, Milkfish runs on the router and reads the SIP headers and rewrites them according to some rules that you set up. It can also redirect a call based on some routing rules that you set up. For example:
"All calls from this server go here, unless they are from this SIP particular address in which case we will send them there."
The web interface allows you to register a homesip account and make calls on that service. To get clever, however, it looks like you need to edit the config file manually. /etc/openser.cfg
Joined: 04 Feb 2007 Posts: 426 Location: Fountain of Youth
Posted: Wed Mar 02, 2011 4:05 Post subject: Working configuration for IPKall
Thank you for the tutorial.
I have it working, however, I did not configure the SIP router to work with SIP (callcentric.com) provider. Now I am scratching my head wondering how to leverage it and what that buys me. I think that the SIP router allows multiple soft \ IP phones to use one IP address, just like multiple computers behind a router. Also the nice features (rules based routing) mentioned by in the tutorial above.
After reading the procedure on the homepage of ipkall.com, it turns out that before you register with IPKall, you need to get a callcentric.com user number. It will be of the format 1777XXXYYYY.
For some reason it is unnecessary to put IPKall's info in the downstream devices (router, softphone): probably because sessions are initiated unidirectionally (in the downstream direction).
Register with IPKall and go fill out the form so it looks like this (what you see is not my phone#):
Note that it is necessary to add in. to callcentric.com => in.callcentric.com
SIP (Session Initiation Protocol) is a plain text protocol, similar to SMTP, in that it is designed to manage the signalling and communications between two SIP devices. http://en.wikipedia.org/wiki/Session_Initiation_Protocol
Like an SMTP email header, you have a source address, destiantion address, and a number of commands, such as INVITE, OK, NOTIFY etc allowing you to exchange information about what you are trying to send and how its going to get there.
The media or payload is not managed by SIP, the same way that you have a MIME attachment in an email with the actual content of what you are trying to send. With SIP, thats handled by RTP in the case of voice and video
You register your SIP client, such as XLite or a Polycom IP phone to a SIP regsitration server. In this case it will be the voiper(2).ipkall.com
You give it your username and password. It says
"You're cool. I now have your IP address to send my SIP stuff to you"
Then someone calls your IPKall landline number. IPKall's server then send the call to your SIP client as a SIP call with all of the SIP information. Your SIP client, responds accordingly and rings, answers and sets up the RTP media path between its self and the IPKAll server.
This is similar to sending you an email, with a PDF attached.
SIP doesn't carry the audio.
If you just want to recieve calls on a single handset or Softphone client, then you dont need Milkfish.
I am an absolute beginner at Milkfish, but I have a good understanding of SIP and have been learning about Kamailio (formally OpenSER, which is what Milkfish is built from), SER and OpenSIPS
Where Milkfish is good, is it allows you to have two or more client connections behind the router on the one SIP account. You can also make SIP calls directly between the clients.
Now, Milkfish runs on the router and reads the SIP headers and rewrites them according to some rules that you set up. It can also redirect a call based on some routing rules that you set up. For example:
"All calls from this server go here, unless they are from this SIP particular address in which case we will send them there."
The web interface allows you to register a homesip account and make calls on that service. To get clever, however, it looks like you need to edit the config file manually. /etc/openser.cfg
Does that help a little ??
Cheers
Chris
You seems to know allot about VoIP.
Im fighting to get voip working with DD-WRT in the first place.
I got the signaling correct like you mention, but not the RTP data. I cannot hear audio.
Some people say there is a big problem with Nat in DD-Wrt and switched to Tomato Wrt and the problem went away.
Can you comment on your VoIP experience with DD-Wrt?
I have a PBX remotely on a datacenter.
The voip softphone installer on my computer can call in and I hear audio and it works just fine.
My hardware voip phone a Siemens, receives calls as well but I dont hear anything.
Same for a linksys and another grandstream phone. They all suffer the same. I forwarded specific ports for each device, unique for each one like 5020, 5030, etc. And it works. But the RTP protocol cannot pass the router.
I noticed on my PBX logs that the softtone actually registers with my public IP and after the call it shows the public IP as the one that send data.
With the hardware devices, they do show the public IP as well when they register, fine, but when I make a call it seems they send their private LAN IP to which my server cant of course send data back.
