Batlabs VoIP Network

k4wtf · Post by **k4wtf** » Thu Jan 22, 2004 6:22 pm

Sundown wrote:
KG6EAQ wrote:
k4wtf wrote:OK folks.

For your amusement, you can now dial "8378" (test) and get to a menu tree that I'll be adding different things to as I get to them.
I didn't get a menu.. I got put on hold then it hung up.
8378... not 2288378

The menu is in the message above. I haven't recorded voice prompts for it yet. You can choose 11# through 16#.

John

KG6EAQ · Post by **KG6EAQ** » Thu Jan 22, 2004 6:31 pm

Sundown wrote:8378... not 2288378

doh!

k4wtf · Post by **k4wtf** » Thu Jan 22, 2004 6:40 pm

OK. I couldn't resist. Feel free to call BAT-SLUT. If someone has an 800 number that you think is more suitable to tie BAT-SLUT to, let me know. I can route the call pretty much anywhere.

John

Sundown · Post by **Sundown** » Thu Jan 22, 2004 6:45 pm

k4wtf wrote:OK. I couldn't resist. Feel free to call BAT-SLUT. If someone has an 800 number that you think is more suitable to tie BAT-SLUT to, let me know. I can route the call pretty much anywhere.

John

Doh!! Should have asked for that number for myself...

Wonder if my girlfriend would see the funny side....

werdnuts · Post by **werdnuts** » Fri Jan 23, 2004 10:38 am

hehe... to think. i was going to ask for BAT-$HIT. i errored on the side of safety and went with CRAP

Post by **alex** » Fri Jan 23, 2004 12:01 pm

Yeah, amusingly enough - all this talk about voice over IP - The telecom guys were at CompUSA installing a mytel system, and earlier this week, our local rep who maintains our phone system came in to discuss voip.

BAT-$HIT would have been quite amusing too.

-Alex

werdnuts · Post by **werdnuts** » Fri Jan 23, 2004 12:56 pm

yeah, alex... cept for the fact that when you type cuss words it replaces them with the little bulged-eye emoticon.... kinda takes the funny away

Post by **alex** » Fri Jan 23, 2004 1:04 pm

Yeah well. We have to keep it somewhat family oriented...

"somewhat" being the key word.

-Alex

k4wtf · Post by **k4wtf** » Mon Jan 26, 2004 11:42 am

Well guys, I had a Cisco 7960G IP phone given to me last night so, we'll be able to test some more with different equipment.

Also, I'm working with the engineering folks at Grandstream on a few firmware issues on their BudgeTone-100 phone. Once those are worked out, I'm going to try to work out a bulk purchase deal with them. We can probably get the phones down into the $50-70 price range.

If any of you have talked to me on the BAT-line, I have been using a BudgeTone-100 phone so, you know what it sounds like and that it does work. (Minor annoyance issues with the firmware is what we're working on.)

I might also be able to get some of the Cisco phones but, they're going to be 3-4 times expensive. (They're AWESOME though!)

Anyway, how many people would be interested in one of the phones? I don't know how many we would have to buy to get bulk pricing but, I need to get an estimate before I even approach them.

Later,

John

Johnny Grep · Post by **Johnny Grep** » Mon Jan 26, 2004 2:30 pm

I'd be interested in Cisco phones - although I prefer Mitels. Can you ask your contact if they can provide us with a price for Mitel 5055's? I can't get any of them here...

k4wtf · Post by **k4wtf** » Mon Jan 26, 2004 2:59 pm

Um, I doubt my Cisco contact is going to get us decent pricing on Mitel phones. They work for Cisco.

John

rmi2065 · Post by **rmi2065** » Mon Jan 26, 2004 3:41 pm

k4wtf wrote:Well guys, I had a Cisco 7960G IP phone given to me last night so, we'll be able to test some more with different equipment.

John, I've tried to configure my 7960 with now luck. I've downloaded the SIP image and flashed the phone, however, when placing a phone call - the call times out. I can place a phone call when using FWD just fine. So, i doubt it is a problem with the phone, firewall or anything in between.

I also have a case (5) of 7910's, i think they only work with CCM. Can someone confirm this for me please. I am willing to part with four of them, gonna keep one for myself. I would like to get $100 per phone. If anyone is interested, let me know. I just might deal on the price.

I'll be online around 4 PM, PST if anyone would like to call about the phones or just call and chat.

k4wtf · Post by **k4wtf** » Mon Jan 26, 2004 3:46 pm

Does the Cisco claim to have authenticated to our sip proxy? If you wanna try now, I can watch the logs and see what happens.

John

rmi2065 · Post by **rmi2065** » Mon Jan 26, 2004 3:51 pm

k4wtf wrote:Does the Cisco claim to have authenticated to our sip proxy? If you wanna try now, I can watch the logs and see what happens.

John

I don't think it does. I be getting home around 4, and then I'll give it a try. I'll also delete the FWD configuration.

rmi2065 · Post by **rmi2065** » Mon Jan 26, 2004 3:52 pm

rmi2065 wrote:
k4wtf wrote:Does the Cisco claim to have authenticated to our sip proxy? If you wanna try now, I can watch the logs and see what happens.

John
I don't think it does. I be getting home around 4, and then I'll give it a try. I'll also delete the FWD configuration.

Actually I'm not sure. It was late last nite when I gave up.

fireradio · Post by **fireradio** » Mon Jan 26, 2004 3:56 pm

rmi2065 wrote:
k4wtf wrote:Well guys, I had a Cisco 7960G IP phone given to me last night so, we'll be able to test some more with different equipment.

John, I've tried to configure my 7960 with now luck. I've downloaded the SIP image and flashed the phone, however, when placing a phone call - the call times out. I can place a phone call when using FWD just fine. So, i doubt it is a problem with the phone, firewall or anything in between.

I'm having the same problem, but with X-Lite. Can't figure out what's wrong. It could be that X-Lite is not authenticating for some reason, but I don't get any error messages at all. It just does - nothing. FWD works fine.

k4wtf · Post by **k4wtf** » Mon Jan 26, 2004 4:35 pm

rmi2065 wrote:
rmi2065 wrote:
k4wtf wrote:Does the Cisco claim to have authenticated to our sip proxy? If you wanna try now, I can watch the logs and see what happens.

John
I don't think it does. I be getting home around 4, and then I'll give it a try. I'll also delete the FWD configuration.
Actually I'm not sure. It was late last nite when I gave up.

Justin,

Your phone is still trying to authenticate it looks like...

REGISTER sip:66.35.64.38 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060
From: sip:[email protected]
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 101 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 3600
Content-Length: 0

9 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.103 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.103:5060
From: sip:[email protected]
To: sip:[email protected];tag=as535bfdd4
Call-ID: [email protected]
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0

It's sending its internal RFC1918 address. That is the problem.

John

rmi2065 · Post by **rmi2065** » Mon Jan 26, 2004 4:36 pm

John,

I'm home now. I'll power up the 7960. Hopefully you'll be able to see something on your end.

rmi2065 · Post by **rmi2065** » Mon Jan 26, 2004 4:39 pm

John,

I'll make the corrections and try again in a few minutes. Thanks

Johnny Grep · Post by **Johnny Grep** » Mon Jan 26, 2004 5:04 pm

k4wtf wrote:Um, I doubt my Cisco contact is going to get us decent pricing on Mitel phones. They work for Cisco.

John

LOL, I misread. Thought they might be a communications distributor.

Johnny Grep · Post by **Johnny Grep** » Mon Jan 26, 2004 5:06 pm

rmi2065 wrote:John,

I'll make the corrections and try again in a few minutes. Thanks

What 7960 FW version do you have?

rmi2065 · Post by **rmi2065** » Mon Jan 26, 2004 5:22 pm

Johnny Grep wrote:
rmi2065 wrote:John,

I'll make the corrections and try again in a few minutes. Thanks
What 7960 FW version do you have?

Application Load ID : POS30202
Boot Load ID : PC030301

I choose an older SIP image just because they are usually less problematic to configure.

I've set the DMZ of my linksys router to the internal IP of the 7960, Ive also set its IP to external. I still not authenticating.

k4wtf · Post by **k4wtf** » Mon Jan 26, 2004 5:32 pm

I saw you authenticate once.

I don't know why some of you are having the problems and some not. Very strange.

John

k4wtf · Post by **k4wtf** » Mon Jan 26, 2004 7:10 pm

OK folks. If you're having problems, or even if you're not, try the following:

Set your sip proxy to: "sipproxy.enterzone.net:5060"

Set your realm to "sipproxy.enterzone.net"

Lets see if this makes a difference.

John

Sundown · Post by **Sundown** » Mon Jan 26, 2004 7:26 pm

rmi2065 wrote:Application Load ID : POS30202
Boot Load ID : PC030301

I choose an older SIP image just because they are usually less problematic to configure.

Running P0S3-06-0-00 here without any problems for several months now.

I've set the DMZ of my linksys router to the internal IP of the 7960, Ive also set its IP to external. I still not authenticating.

Eh? Want to go through that again? At first read, that sounds quite broken...

Also, feel free to send your 7960 config file (if you're using one) through to John and he'll send it through to me for review.

rmi2065 · Post by **rmi2065** » Mon Jan 26, 2004 8:12 pm

Sundown wrote:
rmi2065 wrote:Application Load ID : POS30202
Boot Load ID : PC030301

I choose an older SIP image just because they are usually less problematic to configure.
Running P0S3-06-0-00 here without any problems for several months now.

I've set the DMZ of my linksys router to the internal IP of the 7960, Ive also set its IP to external. I still not authenticating.
Eh? Want to go through that again? At first read, that sounds quite broken...

Also, feel free to send your 7960 config file (if you're using one) through to John and he'll send it through to me for review.

Okay, sorry for the vague post, trying to cook dinner and play on the computer don't mix.

What I meant was - I set up the DMZ in the router to the 7960's NAT'ed address, I the set the phone to look to the public IP address. WHY you ask, well my thinking was this - for some reason John's servers were seeing the NAT'ed address of the phone and didn't know where to send packets. I made sense to me at the time. I thought that if I set the phone to look at the public IP address it would properly route voice packets.

I welcome any constructive input regarding my logic. After all, how better to learn.

k4wtf · Post by **k4wtf** » Mon Jan 26, 2004 8:21 pm

Justin,

You actually authenticated for a few moments earlier tonight. I don't know what you changed since then though.

Here's what I'm seeing from you right now:

Sip read: I>
REGISTER sip:66.35.64.38 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060
From: sip:[email protected]
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 101 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 3600
Content-Length: 0

9 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.103 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.103:5060;received=67.23.67.57
From: sip:[email protected]
To: sip:[email protected];tag=as39c79788
Call-ID: [email protected]
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0

to 67.23.67.57:50558
Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.103:5060;received=67.23.67.57
From: sip:[email protected]
To: sip:[email protected];tag=as39c79788
Call-ID: [email protected]
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Proxy-Authenticate: Digest realm="asterisk", nonce="11949996"
Content-Length: 0

to 67.23.67.57:50558

So, do what I suggested above and change your realm to sipproxy.enterzone.net and your sip proxy to "sipproxy.enterzone.net:5060" or "sipproxy.enterzone.net" and we'll see how that works.

John

Sundown · Post by **Sundown** » Mon Jan 26, 2004 9:06 pm

rmi2065 wrote:What I meant was - I set up the DMZ in the router to the 7960's NAT'ed address, I the set the phone to look to the public IP address. WHY you ask, well my thinking was this - for some reason John's servers were seeing the NAT'ed address of the phone and didn't know where to send packets. I made sense to me at the time. I thought that if I set the phone to look at the public IP address it would properly route voice packets.

OK, having problems with your use of some of the terminology. Firstly, I don't think you want to touch DMZ in the router. Instead, do you know if your router has NAT support for SIP? It doesn't look like it, but it may be an option you need to enable.

Otherwise, try the following:

- Get rid of any DMZ related settings and put the phone back with the rest of the computers.

- Add a static NAT rule in your router to direct packets coming in to the router on UDP port 5060 through to the phone's inside address.

- Set up NAT support in the 7960. From the phone UI, that is Settings -> SIP Configuration , then NAT Enabled = yes and NAT address to be the outside (ie public) address of your router.

Then try and register to John's SIP server and see how that goes.

rmi2065 · Post by **rmi2065** » Tue Jan 27, 2004 6:40 am

John, Sundown

Thanks for your suggestions. When I get home tonight I'll make all the changes and go from there.

Thanks again!

Post by **alex** » Tue Jan 27, 2004 10:47 am

Well,

I've got the VOIP bug, so I went out and aquired two Cisco ATA-186's. Should be here late this week or early next week (yay!).

Hopefully this will make a nice addition to both work and home.

-Alex

rmi2065 · Post by **rmi2065** » Tue Jan 27, 2004 4:02 pm

Holy Cow!!!

It has authenticated with John's server.

I've successfully completed two phone calls, one to TEST and another to BAT-ECHO.

All that I had to do was set up static routing of UDP packets on port 5060 to the internal (NAT'ed) IP of the 7960 and get rid of the DMZ setting for the 7960's IP.

I owe thanks to Sundown for that, and A big thanks to John for providing us with the BAT VoIP network.

So from now on anyone that calls me I will be chatting on my 7960.

Sundown · Post by **Sundown** » Tue Jan 27, 2004 4:33 pm

You just completed your third call as well, but I can't call you back

jhook · Post by **jhook** » Tue Jan 27, 2004 5:06 pm

ok got my phone working last night and now i'm wondering if it's supposed to show who's on line and also who's who and there number's some place thank's Jhook

Post by **alex** » Fri Jan 30, 2004 8:36 am

Anyone have this working with an ATA-186 yet? I aquired 2 of them, and have since been playing for a bit trying to get it working....

-alex

oh:

ata000c30000c2e
Version: v2.15 ata18x (Build 030122a)
MAC: 0.12.48.0.12.46
SerialNumber: INM070666AZ
ProductId: ATA186I1
Features: 0x00000000
HardwareVersion: 0x0006 0x0000

Sundown · Post by **Sundown** » Fri Jan 30, 2004 8:44 am

Yup

You'll probably need to put on SIP firmware though...

Post by **alex** » Fri Jan 30, 2004 8:48 am

Woohoo! There's my aussie friend just when I need him.

As far as I can tell, it's SIP loaded

-Alex

Post by **alex** » Fri Jan 30, 2004 10:00 am

Well - it seems to be working - someone out there try calling if they are bored... 228-9111

-Alex

rmi2065 · Post by **rmi2065** » Fri Jan 30, 2004 5:13 pm

Can someone try and call me. I think I figured out all of my probblems. I just need to see if I am receiving phone calls.

Thanks

k4wtf · Post by **k4wtf** » Sat Jan 31, 2004 5:48 am

I just tried to call you. Didn't go through.

John

BigTex1136 · Post by **BigTex1136** » Sat Jan 31, 2004 6:09 am

OK I have read alot of this VoIP post and mybe this will be a stupid question but what and how do I use this? Will it work as a normal phone to or can you only talk to other people on the network or whatever? Also How much will it cost a month and what will I need to buy or get? Thanks

k4wtf · Post by **k4wtf** » Sat Jan 31, 2004 6:35 am

You can talk to any other person on the BAT network as well as any 800 or other tollfree number. It costs nothing. If you want the ability to receive calls from the PSTN or make PSTN calls outside of tollfree calls, I can do that too but, it is going to cost you money.

John

BigTex1136 · Post by **BigTex1136** » Sat Jan 31, 2004 6:51 am

So how do I get started? What do I need?

k4wtf · Post by **k4wtf** » Sat Jan 31, 2004 7:40 am

You mean you've read this entire thread and you don't know? I'm sorry. If you can't get running after reading this thread, you're going to be a support nightmare and I just don't have the time for that.

If you haven't read this entire thread, go back and read it - COMPLETELY.

John

Johnny Grep · Post by **Johnny Grep** » Wed Feb 04, 2004 8:56 pm

John, how about that 7960 group buy? Any news? Prices?

k4wtf · Post by **k4wtf** » Thu Feb 05, 2004 12:36 am

Nothing yet. I've been swamped lately and haven't had time to look into that. My 7960 should be here at 10:30am so, I'll let you know how that goes.

John

Post by Pj » Thu Feb 05, 2004 6:33 pm

Maybe I missed something somewhere, but where can I change the ring sounds? It says to basically "just change them" but all I see is default, and no provisions to point the software to a wav file on the computer.

k4wtf · Post by **k4wtf** » Thu Feb 05, 2004 10:10 pm

Change them on what? A hard phone or the X-Lite? Not that I know how to do either but, it helps to know what you're trying to do.

BTW: I made some changes to the server config this evening as a result of some things I learned while configuring my 7960. If you were having problems before, try now. If you are having "sudden" problems, let me know.

John

k4wtf · Post by **k4wtf** » Thu Feb 05, 2004 11:50 pm

How cool is THIS?

Bwahahahahaha!

John

KG6EAQ · Post by **KG6EAQ** » Fri Feb 06, 2004 1:28 am

Mmmm XML phone...

Post by Pj » Fri Feb 06, 2004 5:32 am

that would be xlite for the ring/sound question. I got it from the last link you posted.

Batlabs VoIP Network

re phone