FreePbx

[tschak909]

Spoken like a know it all, who understands nothing of the reasons and decisions as to why things were done the way they were. Way to go.

We got rid of FreePBX, because for 99% of what people needed to do, needed to be done in the web admin, under a single user interface, and so that uninformed changing of the system, wouldn't break the telecom part of the system in half. If there is a feature missing from the web admin that you need, please work with us to add it! I'm tired of all this incessant "I don't understand why they did this!" talk. You're acting like a whiny bitch.

-Thom

[gbutters]

sip debug for asp3102 ata ip:
Please note that the spa3102 is serving fro two purposes:
1 - pstn line as a pots trunk - no way no register.
2 - line 1 as a sip extension 204 - this one registers and works.

For the PSTN line setup a phone line in lmce as spa with the phone number (1234567890) as the user and in spa3102 config under pstn line

Line Enable:   yes      
 
SIP Settings
SIP Port:   5061      
 
Proxy and Registration
Proxy:   192.168.80.1
Register:   yes   Make Call Without Reg:   yes
Register Expires:   300   Ans Call Without Reg:   yes
 
Subscriber Information
Display Name:   Unknown  Caller   User ID:   phone number
Password:   password   Use Auth ID:   yes
Auth ID:   phone number   


With those settings the spa3102 pstn line registers for me.

[tschak909]

I am pasting this, because microbrain felt it important to block users private messages, while spamming me with his own:

Your response to my web post today displays the type of attitude that has the attendances to make a person like me not want to return to this website.

You sound like a 15 year old BOY that's still living at home and momma still wiping your ass.

Like anyone else, not fully understanding why the changes were done the way they were, I was hoping for a somewhat reasonable explanation behind these changes. Your explanation would have done well had you left out the condescending, degrading and name calling remarks. Most 15 year old BOYS don't understand that as they haven't lived long enough to learn it.

I was simply trying to make an attempt to help someone whom was beyond frustration to get their problem worked out since it was apparent he wasn't getting much assistance and remarks such as yours not only turn me sour toward you but also the other ton of new people whom may happen across this post.

Where I come from if you can't say something constructive in a somewhat decent attitude where it benefits all to the good then it is usually best to keep your shit to yourself, or, send it to me in a private message so you don't show others your yuppie ignorance, BITCH.

No, you misunderstand, completely understandable, as I was very fried and frazzled after working three contracts in a row for now, and the foreseeable future, and I lashed out.

I am angry, because you've basically not worked with us at ALL over what features you are DESPERATELY needing from FreePBX that can be folded into the Web Admin. Nada, nothing, you just whine and bitch and complain, as do many users on this forum, maybe because they feel entitled to do so, maybe because they feel helpless for some reason. If it's the latter, anyone who has worked with the team directly can attest that we BEND OVER BACKWARDS to help those who help themselves.

This system became a volunteer effort, and the changes that we're trying to make are the result of trying to make the system predictable and maintainable. Neither I, nor foxi352, who made the changes, nor the rest of the development team will step back from these assertions, because we believe that given the REALITY of the development team that we have, the most common use cases used by the telephone system, and the fact that we wanted to present a single user interface for configuring everything, doing what we did was and still is, the best choice.

-Thom

[pw44]

Pw44,

Your problem is within the authentication of your PSTN line., but you know that already.

Three things I would look at: (make note of any changes you make)

1: Did you set the spa for dhcp or assign it a dynamic ip? If you have it as dhcp, change it to static and try that then change the #2 next, if it still don't work put it back.

spa is setted to a fixed ip address 192.168.80.30

2: in the spa under "Proxy and Registration" is it set for yes, if so try to set it to off and then check if it works. If it is set to on it tries to tell LMCE what your spa address is, and since you already assigned an ip you don't need to register.

Proxy and Registration: Register is set to YES. I will give a try with NO.

3: as much as I hate to say this, go back to LMCE 8.10 because the current 10.04 will not work without the ability to setup this box with something like (here it comes, wait, wait,) FREEPBX...... and don't feel too bad as you won't be the only one having to do it.

I myself was going to try and set LMCE up to be an extension of my asterisk box, but after studying how LMCE communicates with asterisk I can't do that either nor can I make the asterisk box work as a in/out trunk to the LMCE because of the changes they made in how the LMCE deals with asterisk. I'm still not sure if I fully understand why the powers to be had to change up asterisk/lmce. If it truly was an issue (as I have read on some other post) that too many people were breaking LMCE using FreePbx I would have thought it would have been easier to just make FreePBX an add-on and not offered help when they screwed it up, just my opinion.

microbrain

Yes, would be prefereable do have something like freepbx, but it's gone, so learning all again from ground zero (where and what, beside the criptic syntax, which was hidden by freepbx). But, as said, it's gone.

Thx for your trying to help Let's see if we get it, mostly by trial and error, due lack of documentation.

[pw44]

For the PSTN line setup a phone line in lmce as spa with the phone number (1234567890) as the user and in spa3102 config under pstn line

Line Enable:   yes      
 
SIP Settings
SIP Port:   5061      
 
Proxy and Registration
Proxy:   192.168.80.1
Register:   yes   Make Call Without Reg:   yes
Register Expires:   300   Ans Call Without Reg:   yes
 
Subscriber Information
Display Name:   Unknown  Caller   User ID:   phone number
Password:   password   Use Auth ID:   yes
Auth ID:   phone number   


With those settings the spa3102 pstn line registers for me.

Gbutters,

thank you for your email. I will try this configuration later and will report back with results..

Best regards,

Paulo

[pw44]

I'm not blaming not having freepbx, but the lack of documentation and information, and that's not your fault. I know it's a volunteer effort and i also understand it's not easy to make it easy for end users.
I would gladly help, if i had the time and the knowledge for it. Some very little contribution i gave (fail2ban, and some voip trunk settings).  

Well, if i can suggest, how about a little tutorial about, comparing the freepbx settings (trunk - peer detail, user detail), outbound route, dialplan according to trunk and register, or a hidden panel where we could tune it.

For me, at least following directives are missing.
* 83    0    18    0    sip.conf    general    alwaysauthreject    yes
* 85    0    18    0    sip.conf    general    nat                            yes
* 86    0    60    0    sip.conf    general    externhost            mydyndns.homeunix.org
* 87    0    5    0    sip.conf    general    externrefresh            5
* 88    0    60    0    sip.conf    general    localnet                   192.168.80.0/255.255.255.0
* 89    0    9    0    sip.conf    general    allow                    g729
* 90    0    10    0    sip.conf    general    allow                    g723
* 91    0    101    0    sip.conf    general    register                    pwollny:XXXXXXXXX@sip.voipcheap.com/2062036594


I don't know if i did it right, because i could not find what are cat_metric and var_metric for.
I'm not asking to have someone doing it for me, but i wish to know where to put what i need, and i'm not finding out

Best regards to all,

Paulo

[Marie.O]

microbrain,

I wonder what you did to your other asterisk system. It is the exact setup I have here. My main asterisk acts as my ISDN gateway and caters to two offices, and is the outgoing line for my LinuxMCE system. Just added the phoneline details and it worked.

[pw44]

For the PSTN line setup a phone line in lmce as spa with the phone number (1234567890) as the user and in spa3102 config under pstn line

Line Enable:   yes      
 
SIP Settings
SIP Port:   5061      
 
Proxy and Registration
Proxy:   192.168.80.1
Register:   yes   Make Call Without Reg:   yes
Register Expires:   300   Ans Call Without Reg:   yes
  
Subscriber Information
Display Name:   Unknown  Caller   User ID:   phone number
Password:   password   Use Auth ID:   yes
Auth ID:   phone number   


With those settings the spa3102 pstn line registers for me.

Thank you. It worked. spa3102 registered. This is a difference between the old asterisk (used in release 8.10) and the new one (used in release 10.04).

Again, big THX!!!!!!!!!!!!

Wiki updated with the info.

[Marie.O]

pw44,

instead of posting it to the wiki, see if you can create a pnp script for the SPA. Look at the grandstream perl scripts how it is done.

[pw44]

Ja, mein Kommandant! Werde da nachschauen!

[Marie.O]

[microbrain]

posde,

I didn't do anything to my asterisk server, everything on it is and has been working fine for several years without a glich. So lets just accept I'm totally ignorant when it comes to LMCE and start at the beginning.

Since I don't find any info on how to setup 10.04 phonelines maybe you are willing to share how you are connecting to your asterisk box, with sip or aix2?

My original set up (8.10) was a peer/user arrangement using sip that worked just fine. I've tried to set it up as a  "peer asterisk box as an extension" and as a "fried/friend arrangement" and neither is working. I'm not even seeing the LMCE box trying to communicate with the asterisk box through the sip debug on my asterisk box, however I am seeing on the LMCE box the asterisk box trying to communicate with the LMCE box.

So that I may get a better understanding of how you are interfacing the two separate boxes together could you post your asterisk trunk details (without passwords of course) and your LMCE phoneline info you entered into LMCE 10.04 and maybe some info on how LMCE handles the peer details and codecs.

Thank you sir

microbrain

[Marie.O]

SIP. I merely entered username, password, phonenumber, prefix and host address.

My main asterisk's SIP conf for the LinuxMCE "user" looks like this:

Code: [Select]

[959]
type=friend                     
context=privat          ; Where to start in the dialplan when this phone calls
host=dynamic            ; we have a static but private IP address
defaultip=10.1.2.67
insecure=port,invite
                                ; No registration allowed
nat=no                          ; there is not NAT between phone and Asterisk
canreinvite=yes         ; allow RTP voice traffic to bypass Asterisk
dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
username=959
secret=myverysecretpassword



[microbrain]

Under "PhoneLines" there is a "Status", is this status showing it's registered, in use, what?

The problem is that the line shows "enabled" but status shows "no". I'm assuming (we know what that means) that the "status" means it's not registering with the other end.

I can call from my asterisk box into LMCE and it rings the extension assigned on incoming route, I can not though dial out to the asterisk box as it continues to show "congestion/busy".

What version of asterisk does LMCE use?

Thanks
microbrain

[microbrain]

Well I got the "phonelines" as it's call set up and figured out that "Status" is for show if registered or not (no).

All works but audio. What codec is assigned within the LMCE when a phoneline is setup?

In reference to possibly making certain available options for the end user to access, it would sure be nice to have the following (just a request -- not a demand) so as to make it easier to set up trunks and outgoing dial patterns:

Outbound Routes Setting (for setting up different dial patterns based on best price practice)
Outbound Trunk Dial Rules (The current "prefix" lets you select the trunk to use but not dial rules on that trunk)
Outgoing Peer Detail Settings ( allows for fine tuning and or manipulation of trunk/peer info)
Incoming Peer Detail Setting ( same as outgoing peer details)

Since LMCE sip.conf info is stored within the database and not accessible under /etc/asterisk (where it normally resides on most asterisk distros) it makes it hard to modify trunk & extension details along with codecs when necessary. Although I can set all this in my separate asterisk box for someone who doesn't have a separate box and needs to choose different dial patterns on outbound routes or trunks and/or special mods to peer info I could see where this would be beneficial. Just a thought.

Thanks
microbrain