FreePbx

[microbrain]

I downloaded the stable addition of 8.10 and installed it on a test box to try and get a grip on an actual working LMCE. In playing with it I noticed that FreePBX was a part of that version. I know that I can download and install FreePBX but I'm not sure if the current version of FreePBX is what works with 8.10 or 10.04-26551. Is there a script, add-on a known way that I can install FreePBX on 26551 of 10.04 and where would I need to change the link in LMCE admin page to get to FreePBX once I install it? Maybe someone has install it on 10.04 and would like to post how they did it. I read in some other posts that it was removed due to people not familiar with FreePBX making changes that caused breaking LMCE.

I would like to take advantage of additional options within the Asterisk PBX that are not accessible within LMCE.

Thanks

microbrain

[maverick0815]

As far as I know freepbx is no longer part of 10.04 and can't be used anymore. Asterisk is beeing configured via the database.

[posde]

maverick is correct. We no longer use freepbx. We have our own web frontend which manages phone lines, phones, routing and stuff. If there is anything missing from our web frontend, let us know.

[pw44]

How about a front end takes looks like freepbx, with sections divided, would make live much easier..... there are too many variables and a very simple interface, or better, a no interface make it all a PITA. I'm fighting with it for some weeks, and no results. No even ONE trunk is working, and a had THREE working with 8.10, all configured with freepbx and working like a charm. So, IMHO, it may be very good having asterisk with realtime database, but there is almost no documentation about, and worst, i could not find anyone using it to get a help and discuss how to make my old configs work with this combo.

[posde]

pw44,

there are people who just use the web admin, fill out the forms, and have multiple lines working. If you have problems with the web frontend, please detail them.

[pw44]

Posde,
1 - Linksys SPA-3102 - no way to make it register, and so, no landline.
2 - voipcheap -  needed to add SIP_ANON yes - calling my DID number from my mobile, my home phone rings, outgoing sound works (i hear on the mobile), incoming voice (from mobile) cannot be heared on the home phone. Calling from home to outside, it takes almost 5 minutes to calling phone (my mobile) to ring, and no sound in or out.
3 - sipgate.de - does not register. Nothing happens.
All three trunks were working on the 8.10 release. Hardware is the same: core/hybrid, cisco phone, linksys spa-3102, router, adsl modem, all exactly the same. The only change was from 8.10 to 10.04.
So, i'm digging for the last 40 days for a solution, but no way to make it work. Too less documentation about. I'm out of knowledge to find out what is wrong, as all worked before.
TIA,
Paulo

[microbrain]

From what I am observing with LMCE 10.04 and phone line/phones set-up I can see where pw44 is justly frustrated beyond belief.

As it is now if you happen to have a Grandstream or Snome phone that does an FTP boot then setting up a plug and play sip phone probably works well for the common user. But, for someone like pw44 or myself whom has a ATA setting up this box to work can be a hassle unless you know the inner workings (code) of Asterisk.

There are some VoIp providers that you just have to go in and modify certain details of the trunk set-up and by having something like a FreePBX front-end to Asterisk makes this a breeze. Things like Outbound CID, maximum channels and dial rules, user details and peer details not to mention the registration string. In addition, LMCE now does not allow for modifications of certain details of an extension when setting it up. Such as what codecs to allow or deny, what Ip's allow/deny, or if it is a SIP, IAX2, DAHDI, ZAP or Custom device or nat settings.. And, for those of us that are knowledgeable, setting up IVR and voice mail options that meet our needs for around the home.

What do you do in the case of a person who has their own landline, wants to install an FXO/FXS card into the LMCE box and use sip phones through out the house? Just asking for personal knowledge.

I would think, IMHO, that a downloadable add-on GUI like FreePBX or something similar would have been the way to go that would allow those of us knowledgeable a way to set-up Asterisk to meet our needs, or, just setting LMCE code to connect to an external Asterisk Box. Asterisk is like LMCE, it's free and setting up a separate machine is less costly then setting up  a LMCE machine and probably would have made LMCE coders a lot happier not having to mess with the Asterisk portion.

For pw44, he has what sounds like three issues. One, the trunk detail to his SIP provider and two possibly a NAT issue (one way audio is normally caused by NAT issues), and three - as  far as the SPA-3102, if the parameters within it are set properly it should register with the LMCE if not then he needs to check its parameters. For the SPA-3102 he can determine what's going on by running sip debug command on a command line entry on the main server and watch what is going on when it tries to register.

Since I haven't learned all there is to LMCE, the question I have again is is it possible to download FreePBX and install it without totally destroying LMCE? Has anyone tried to do it - is it possible? If not then I guess those of us that need the ability to have Asterisk do what we wish and keep it part of the LMCE idea then people like me will be forever stuck with LMCE 8.10 unless something else comes along.

Asterisk has a lot to offer even the home user or better still the SMB. Not taking advantage of its potential I think leaves LMCE limited in its potential, again just my opinion.

microbrain

[posde]

pw44,

if your three phonelines did work before out of the box, and they no longer work out of the box, please open a bug ticket.

if your three phonelines did not work before out of the box, and you want them to work again in 1004, please open a feature patch ticket, detailing what you had to do in 810 to get them to work.

[pw44]

So, before beginning to blame the new asterisk combo, i inserted the following in the database:
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
Also did set SIP_ANON yes.
As as i told before: All three trunks were working, with trunk selection by dialplan, without glitches.
I'm not an asterisk guru, did research a lot in order to make it work, to understand how to build the dialpan and to have all working, and now all i learned is worth zero.
I don't know if it's a bug, if it's a configuration issue, i'm not finding out how to proceed with config. Is there any at least medium documentation where i can find help? I could not find any.
That's all.

[posde]

pw44,

could you answer the implied questions of my previous post, please

[pw44]

Answering the question: the three lines worked on my 8.10 release.
I will open a bug ticket.
Thx.

[posde]

Did the work ootb or did they work with manual fiddling?

[pw44]

On 8.10 nothing worked for me ootb, all trunks needed to be feeded, because sipgate, voipcheap and spa-3102 did not had the amp_create****, and later i did create amp_create_sipgate and amp_create_voipcheap (and created wiki for it). spa-3102 was manually created, so as the dialplans.

[posde]

pw44,

than I would not call this a bug, but a feature request, please. Could you detail what settings were needed to get the phone lines working, please? And if possible, do one for each, that way foxy, or whoever is going to look at it, can tackle one at a time, and put things forward. A working sip.conf for each of the phone lines would be a good thing to attach to the tickets.

[pw44]

Posde,
thx for answering:

I would not say it's a feature request, because is expected that any user would be able to have at least one trunk working with the sip provider of choice . Ok, not all sip providers are supported by the devel group, so, some documentation should be provided, and making the trunk work would add new providers. I did it with sipgate and voipcheap, but i had something to research and digg. spa3102 was done by Seth.
 My 8.10 working config.
Trunks: SPA-3102 - the spa config is the same as found in: http://wiki.linuxmce.org/index.php/Linksys_SPA3102
            Voipcheap wiki: http://wiki.linuxmce.org/index.php/VoIP_with_voipscheap.com
            sipgate.de wiki: i remember that i created it, but it's not there

Well, to the asterisk confi files. If the freepbx version is needed, please let me know.

The sip.conf from the working config:

Code:

[general]
#include sip_general_additional.conf

bindport = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
alwaysauthreject=yes ; required by fail2ban
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes

; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf

sip_additional.conf

Code:

[sip.voipcheap.com]
type=friend
qualify=yes
insecure=invite,port
host=sip.voipcheap.com
dtmfmode=auto
disallow=all
context=from-pstn
allow=ulaw
allow=alaw
allow=g729

[sipgate]
username=username
type=peer
secret=xxxxxxxxxxxxxxxxxxx
qualify=yes
port=5060
nat=yes
insecure=invite,port
host=sipgate.de
fromuser=username
fromdomain=sipgate.de
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
authuser=username
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729
call-limit=50

[sipgate_de]
username=username
type=friend
secret=xxxxxxxxxxxxxxxxxx
qualify=yes
port=5060
insecure=invite,port
host=sipgate.de
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729

[spa3102]
username=spa3102
type=friend
secret=lmce
qualify=yes
port=5061
nat=never
incominglimit=1
host=dynamic
dtmfmode=auto
context=from-trunk
canreinvite=no
allow=ulaw
call-limit=50

[voipcheap]
username=username
type=friend
sendrpid=yes
secret=xxxxxxxxxxxxxxx
qualify=yes
port=5060
nat=yes
insecure=invite,port
host=sip.voipcheap.com
fromuser=username
fromdomain=sip.voipcheap.com
dtmfmode=auto
disallow=all
context=from-pstn
canreinvite=yes
authuser=username
allow=ulaw
allow=ulaw
allow=g729
call-limit=50

sip_registrations.conf

Code:

register=usname:xxxxxxxxxx@sip.voipcheap.com/2062036594
register=username:xxxxxxxxxx@sipgate.de/054138594676

sip_nat.conf

Code:

nat=yes
externip=myhostdyndns.homeunix.org
externrefresh=10
localnet=192.168.80.0/255.255.255.0

localprefixes.conf

Code:

[trunk-4]
rule1=00+XXXXXXX.

[trunk-2]
rule1=XXXXXXXX
rule2=08+08|00XXXXX.
rule3=005521|XXXXXXXX
rule4=031+0055|XXXXXXXXXX
rule5=031+0|XXXXXXXXXX
rule6=031+XXXXXXXXXX
rule7=031+011XXXXXXXXX

[trunk-3]
rule1=00+XXXXXXX.

If there is any additional configuration file you need, please let me know.

TIA,

Paulo