Dang, I was hoping that would

Christopher Taylor · ‎07-04-2014

To start, I'm running my own asterisk server and have setup multiple sip softphones and a POTS to IP adapter with sip.

I recently got a real IP phone to hook up to my phone system, and have set it up with the sip firmware thanks to the community. That post is HERE.

The problem that I have run into is not knowing how to get it to register with a sip username and password. I can't figure out where to go to input the name and password for the sip setup. Every other device I've setup on my phone server had a clear indication of where to put the UN/PW.

Thanks everyone!

For an example of how things look when I set them up before:

Leo Laohoo · ‎07-04-2014

Ok, I've never played with this router before ... and this is no Cisco stuff. So I'll try to "interpret" this as much as I can.

Your Primary SIP Server is incorrect. This should be your VSP's (voice service provider) IP address. Depending on the VSP configuration but the Primary SIP Server and the Outbound Proxy is usually the same.

SIP User ID and Authenticate ID should be your username. Authenticate Password should be your password.

Talk to your VSP and ask them who is doing the NAT. If the VSP are doing NAT then you need to disable the NAT from your router to your VSP or you'll get one-way audio.

Another thing to ask your VSP is whether or not they will support ALG. If they do support, then leave it ALG on. If not, turn it off.

Wait a second ... This is not going to work. What is your setup???? In my opinion, what you are setting up is making your modem act like a Analogue Telephone Adaptor. Basically, you plug your analogue phone to your modem and make VoIP calls.

But if you are dealing with a Cisco VoIP phone already, this is not the right way of doing things. These information should be in the SEPmacaddress.xml.cnf file.

If you are running Asterisk (what flavour?), then the first thing you need to do is get your Asterisk to talk to your VSP. Once you get this running, you get your phones to talk to Asterisk.

You need to get the settings from your VSP because this is what will make-or-break your settings. This holds true with ALG and NAT. You need to get this sorted out first before you start touching the Asterisk.

Christopher Taylor · ‎07-04-2014

Woah, sorry for not specifying a couple of things. I had typed up a better question, but then it all got deleted so I wrote it up real quick again and accidentally left a few things out. I'll start from the top and try not to leave anything out.

1. I own a Rasberry Pi computer that I had installed asterisk on it. (RasPBX version). This server is hooked up directly to my router (a Linksys E2500). This PBX server holds the IP Address 192.168.1.17.

2. I have a POTS phone that runs to an adapter. This adapter is the Grandstream device. (the image above). It also runs to my router.

3. I have the extension "42" and a password setup in my PBX server, which is then put into the section you see in the screenshot. This has all been working for about 6 months now.

4. I also have some PC's that run Zoiper, when l setup those it was similar to the Grandstream adapter. All you had to do was provide the server host, username (extension), and password. They all work beautifully.

5. I received the new CP-6961 IP Phone as a gift to hook up to my phone server. I installed the sip firmware as you so kindly helped me out before, and I was hoping that it would provide a menu like something I would have seen before, where I could put the host, extension, and password. Unfortunately this is not the case.

I was pretty sure is was something I had to setup in the "SEPmacaddress.cnf.xml" file but could not find any examples of how it should look online.

Thank you so much for your help, and I apologize for the lack of clarity in my original post.

EDIT: One more thing. I don't have a VSP. I route my google voice calls through my PBX.

Leo Laohoo · ‎07-04-2014

I was hoping that it would provide a menu like something I would have seen before, where I could put the host, extension, and password.

Your Asterisk is your voice gateway. Your Asterisk speaks to your VSP. Your phones and ATA can talk to your Asterisk and your Asterisk can assign extension numbers to them. But first you need to get your Asterisk to your VSP. This is the most important part of the the process. Mixing how to get your phones to talk to your Asterisk to your VSP is confusing.

You need to talk and ask your VSP the following questions:

1. Who will do the NAT-ing?

2. Does your VSP support ALG?

I was pretty sure is was something I had to setup in the "SEPmacaddress.cnf.xml" file but could not find any examples of how it should look online.

A working example is found HERE.

I have two 7960 and a 7970 talking to an Asterisk/FreePBX running on a Raspberry Pi since June 2013. Working fine. I can make calls, transfer calls, call extension, etc.

Christopher Taylor · ‎07-05-2014

This all mostly makes sense, except I don't have a VSP.

The only thing I'm not sure about is where exactly I put the username and password for the sip extension I setup in the "SEPmacaddress.cnf.xml" file.

Thanks for all your help so far.

Leo Laohoo · ‎07-05-2014

The only thing I'm not sure about is where exactly I put the username and password for the sip extension I setup in the "SEPmacaddress.cnf.xml" file.

I presume you want your phone to talk to Asterisk using an extension number 112233 which you set in Asterisk. Your voice-mail number is 999.

If you go to the "dalrae" website, scroll down to the column that says "<sipLines>"

<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>Chris' Phone</featureLabel>
<name>112233</name>
<displayName>112233</displayName>
<contact>112233</contact>
<proxy>ASTERISK_IP_ADDRESS</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>112233</authName>
<authPassword>PASSWORD</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>999</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>

Do you have a dialplan.xml file and XMLDefault.cnf.xml file?

Go HERE for more detailed instructions on how to set and create them.

Christopher Taylor · ‎07-05-2014

Very strange, this is exactly what I thought it was supposed to be.

I set it up in my PBX like I do my other phones and setup my .xml like this:

</autoAnswer>

<sharedLine>false</sharedLine>

<callerNumber>false</callerNumber>

<redirectedNumber>false</redirectedNumber>

</forwardCallInfoDisplay>

</line>

Everything is setup like so, but after it pulls the files, and I reboot the phone, it just comes up and says "Phone not registered".

I then tried setuping up the second extension and that didn't work either. Do you think it might because there is the "CallManger" section in it that doesn't actually point to anything? Thanks for all your help again. Hopefully we can get this figured out. I'm going to keep playing with it.

EDIT: And I know that it's pulling the information fine from my TFTP server because when I pull up the log it's local ip I get this:

The phone DN's are the extensions I setup.

Leo Laohoo · ‎07-06-2014

Do you think it might because there is the "CallManger" section in it that doesn't actually point to anything?

Try putting the IP address of your Asterisk.

Also, look under "<natEnabled>" and try turning the stuff on/off.

Christopher Taylor · ‎07-06-2014

Dang, I did both of these things already, and combinations of both. I even changed the timezone and time server to make sure it wasn't that. Hmmmmm.

Here is my current .cnf.xml file. Do you see anything wrong with it? EDIT Here: http://pastebin.com/rsa55jfy

Leo Laohoo · ‎07-06-2014

Can you run a debug on the Asterisk?

Christopher Taylor · ‎07-06-2014

Here is the sip debug from asterisk. http://pastebin.com/N9QNEQd8

I looked through it and I don't see anything from 192.168.1.107 (cisco phone).

Thank you again for taking your time to look into this issue for me. I really appreciate it.

EDIT: Here is a section of the console log I pulled from the phone. http://pastebin.com/mjkNPcyg

It shows (OK) next to CUCM. This address is the Asterisk Server, do you think that means it was able to connect to it? If so, I wonder why it won't authenticate.

Leo Laohoo · ‎07-06-2014

I can't see your phone hitting Asterisk. I can see something talking to Asterisk from a public address of "24.155.139.69".

Christopher Taylor · ‎07-06-2014

Yes, that is my external ip address, it's showing up because I have a softphone setup outside my network to test how well it works over VPN. I just disconnected it as it was just a test line. That way it won't show up in logs in the future.

I wonder why it's not hitting the asterisk server.

1. The IP and the un/pw are correct.

2. It's pulling the configs from my TFPT server.

3. I've turned NAT on and off.

4. Changed callmanger IP address.

5. Quadruple checked that everything was setup correctly in asterisk.

6. I also tried changing the listen port in asterisk and in the phone config.

When I had the NAT server turned on, I was using my external IP address, is this the correct thing to do? I also tried using my asterisk server IP address.

Thanks again!

Leo Laohoo · ‎07-07-2014

The Asterisk and the phone are in the same IP subnet?

Christopher Taylor · ‎07-07-2014

Dang, I was hoping that would be the problem, but they're both under the subnet mask of 255.255.255.0

Register sip phone without callmanager