do you think it is possible??

Hi Denis,

You have my sympathy!

Someone with decent amount of experience probably _might_ be able pull it off but if you are looking for a reality check, I think that trying to get what you are looking to do in 2 days starting from scratch with minimal linux/asterisk is highly likely to end up unsuccessful and your personal situation will be even worse.

I recognize that you and your company are in a real jam at this point, but the only way you can expect a successful outcome using an asterisk based solution is going to be to get more time [like two weeks instead of two days] or pay someone to do most of the work for you.

I am curios , did you try your system before deploying it? I am not familiar with the system you mention.

Luckily you landed in the right place, plenty of good folks here that will help you get this running and the tribox build is rock solid.

I am trying to understand how one gets to the point of desperation that they are trying to install a phone system for 75 users on an a community supported solution over a weekend.

If you want to succeed here is my suggestion:

1) Obtain a system with on the supported hardware list
2) I hope you have supported cards for any PSTN interfaces you have
3) Load trixbox CE on the system
4) Enable remote access

Now #5 post a bounty in the bounty forum. I would suggest a minimum of $1000.00 then look carefully at posts by the persons that respond to your offer and choose the one you feel most comfortable with.

If somebody is willing to work for significantly less than a grand under these circumstances I would be very critical of entrusting my business to that individual.

Scott

So, if I'm understanding correctly, this Patton device connects to the 3cx server via ethernet? Does it create a SIP trunk?

If that's the case, it should just be a matter of getting the trunk set up on trixbox, then pointing all your phones to the trixbox.

If I were you I'd start by getting a vmware copy of trixbox and trying to set up the trunk. If that goes well then you can start planning your inbound and outbound routes in that vmware session and dedicating a couple of phones to it to test. Once you have it going the way you want, then you can install CE on your hardware (I'd suggest using a different HD if possible, then you can just pop the old one back in if you have to to revert) and then migrating your settings. If you get stuck at specific points just post back here and someone will probably try to help if they can.

I wish you luck, that's a tough position to be in. (Just out of curiosity, you mentioned 3cx on Windows XP -- is that how you're running it?)

sounds like there is a memory or file descriptor leak of some sort in the software that you are using.

a) if you only have 75 extensions, the system should "ROCK", but, obviously, it isn't. :)
b) look for the book (online, free pdf version), "trixbox without tears, v2", read through it, skipping the advanced stuff like, connecting two trixbox's together.
c) grab a 1 GHz or better PC with at least 512 MB Ram and 20GB or larger HD.
d) install trixbox on it
e) configure phones against it. (so that you can do extension to extension dialing verified)
f) Trunk it up with a paypal friendly carrier like Teliax.com or carriers.icall.net (this way you can get a temporary DID for inbound, and also get origination for testing purposes, you can always cancel it, no obligation type thingies they are)
g) Presuming F works good, then trunk it up with the Patton and connect all your phones to it.
h) Load test the software? don't know how to simulation the traffic your are describing.
i) Redo all the above on the real hardware after being satisfied that the configuration will work.

Good Luck!

Have not used AsteriskNow, but do like I suggested with a PC and TrixBox CE, the install is automatic, just login as root when done so that it will tell you the web page to visit to configure it.

There are also other FreePBX engine based install kits available.

Without more specific questions, I'm not sure what more I can tell you over what I already have.

Edit: nevermind. I had the reply window open for a long time while I was doing other stuff, and in the meantime many other things were posted.

http://dumbme.voipeye.com.au/trixbox2/trixbox2_without_tears.pdf

FYI: Don't Upgrade firmware on phones arbitrarily! You will regret if you do!

since they are in my did range (600-699). manual says redifinable but not how

but your post will keep me going. thank you very much Scott for your support. Here is the running config of my patton

#----------------------------------------------------------------#
# #
# SN4960/1E30V/UI #
# R4.1 2007-05-24 H323 RBS SIP #
# 2001-01-18T18:28:58 #
# Generated configuration file #
# #
#----------------------------------------------------------------#

cli version 3.20
webserver port 80 language en

system

ic voice 0

system
clock-source 1 e1t1 0 0

profile ppp default

profile call-progress-tone defaultDialtone
play 1 1000 425 0

profile call-progress-tone defaultBusytone
play 1 480 425 -7
pause 2 480

profile call-progress-tone defaultReleasetone
play 1 240 425 -7
pause 2 240

profile call-progress-tone defaultCongestiontone
play 1 240 425 -7
pause 2 240

profile tone-set default

profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_WAN
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

interface IF_IP_LAN
ipaddress 192.168.0.7 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context cs switch

routing-table called-e164 RT_ISDN_TO_SIP_0
route T2 dest-interface IF_SIP_0 MP_REM_CLIR

routing-table called-e164 RT_SIP_TO_ISDN_0
route default dest-interface IF_ISDN_0

mapping-table calling-pi to calling-e164 MP_REM_CLIR
map restricted to ""

interface isdn IF_ISDN_0
route call dest-table RT_ISDN_TO_SIP_0

interface sip IF_SIP_0
bind gateway GW_SIP_0
service default
route call dest-table RT_SIP_TO_ISDN_0
remote-party-id called-party
address-translation outgoing-call request-uri user-part fix 10000 host-part to-header target-param none
address-translation incoming-call called-e164 request-uri

context cs switch
no shutdown

gateway sip GW_SIP_0
bind interface IF_IP_LAN router

service default
domain 192.168.0.20
defaultserver manual 192.168.0.20 loose-router
registration manual 192.168.0.20
user 10000 authenticate password LbFCDNv4/Fk= encrypted default register

gateway sip GW_SIP_0
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_WAN router
no shutdown

port ethernet 0 1
medium auto
encapsulation ip
bind interface IF_IP_LAN router
no shutdown

port e1t1 0 0
port-type e1
clock auto
framing crc4
encapsulation q921

q921
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
bchan-number-order ascending
encapsulation cc-isdn
bind interface IF_ISDN_0 switch

port e1t1 0 0
no shutdown

Its auto generated by a template provided by 3cx. Its the same i found in this italian easyasterisk site

http://www.easyasterisk.it/modules/newbb/viewtopic.php?topic_id=2...

Google translates it as follow:
Salve.
ho un gateway voip paton 4960 che mi interfaccia con una PRI E1. I have a Gateway 4960 voip paton I interface with a E1 PRI.
qualcuno ha idea su come configurare il trunk sip di easyasterisk e magari anche come configurare il gateway patton 4960? Someone has ideas about how to configure the trunk sip of easyasterisk and maybe even how to configure the gateway patton 4960?
Grazie arrivederci. Thanks goodbye.

And the reply says
Based on various tests, I found the functional configuration.
Se dovesse servire a qualcuno: If someone were to serve:
Lato easyasterisk: Easyasterisk side:

[patton] [Patton]
host=192.168.100.51 Host = 192.168.100.51
type=friend Type = Friend
dtmfmode=rfc2833 Dtmfmode = rfc2833
canreinvite=no Canreinvite = no
fromdomain=netsystems.it Fromdomain = netsystems.it
insecure=very Very insecure =
context=incoming Context incoming =

lato patton 4960: 4960 Patton side:

webserver port 80 language en Port webserver 80 en Language

system System

ic voice 0 IC voice 0
low-bitrate-codec g729 Low-bit rate codec-g729

system System
.
.
.
.
.
continuing the same config mine has. This gave me some hope that patton would be just translating the [patton] portion (which is not a patton config and i presume belongs to a config or ini file of this easyasterisk).

Yes My E1 is currently used for anything else but the internal calls. All outgoing calls are pretended by 0 (and the number, what ever it is 2-9digits or 00-digits for international or 69-8digits for mobiles etc etc) and routed there. And Yes since e1 provider has the company numbers serves and incoming traffic.
All extensions are 3 digit same with DID (600-799) and can directly reached from outside. Head numbers which are dived in three groups (two or three each group) correspond to three different brand names (belonging to same company) and are answered by secretary for one of them (different policy) and call assistants for the other two (2 menus each greek-english). Answered calls by assistants are forwarded to call groups accordingly (button pressed). Secretary can forward to extensions or groups. If a group doesn't answer for some reason call will endup to secretary no matter if where originally answered by her or assistant. Calls not answered by extensions (users over here have a dislike on putting calls on hold and picking others) endup to secretary also. She decides what to do with them eventually accordingly (forward them again to extension or groups or ask to call later so she needs good clear view of extension status).

I went through freepbx setup pages and i cannot find any follow me settings or ring groups as in v2notears pdf. Is it CE or just my module core 2.4.0.0 that needs upgrading. I think that without them i cannot implement my plan.

Where do i paste your trunk config? In outgoing / incoming settings under "add sip trunk" or some "config edit" file.

Unfortunately i am curently at home installing a box. I will go to work with new server for further configuration an testing (after some sleep you see its saturday morning here and i am on this since yestarday). I know time difference will be a problem so i am trying to get the most of info so that i have thinks to do. I am very worried if restricted options under inbound call control its a matter of trixbox version, freepbx module version or something else. Under inbound call control appears only two choices Inbound routes and zap channel DIDs.

Ok, I am still here. You need to install the follow me and ring group modules from FreePBX.

Just click on "check for updates online" and then install the modules.

There should be a ton of articles on configuring the Grandstreams by hand on this site.

You can put the username and password in the trunk config if you don't want to change the password.

Scott

Ok, last post for the night. Trunk config goes in the trunks under peer settings settings.

The authentication syntax is simple if you leave authentication on the patton just add:

username=
secret=

If you check the Anonymous SIP box I described previously you should get a trixbox message when you dial a DID. If you get that you are very close to having it working.

Scott

What exactly are you doing on?

Do you have a trunk built to the Patton?
Is it registering?

from the CLI please send the output, in code tags for the following commands:

Trunk section of /etc/sip_additional.conf

go int asterisk cli
asterisk -r
send output of command sip show peers.

Scott

and i am as happy as a lotto winner. I think i should go for did numders now and incoming routes to test incoming traffic.

this is all sip_additional.conf has for patton (user 10000)

[10000]
secret=10000
type=user
context=from-trunk

and at the end

[patton]
host=192.168.0.7
username=10000
secret=10000
type=peer

this is where i should put the additional stuff you've send yesterday?
allow=ulaw
canreinvite=no
disallow=all
dtmfmode=inband
host=192.168.0.7
insecure=very
port=5060
qualify=yes
type=peer

dont really know how to access asterisk cli (its differrent than os cli i presume)

If you can make outgoing calls do not add anything to your trunk_config.

FreePBX writes that file, that is why I wanted to look at it when you where stuck.

Add an inbound route. You can watch the debug output and see your CLI information if you get stuck. One route per DID.

One more thing, if you do a sip show channels when a cal is up it should be alaw not ulaw in Europe.

You are in good shape now.

what do you have in your trunk settings? (via the web page).

I have an unauthenticated sip peer I use: enclosed is for comparison:

Outbound Caller ID:
Trunk Name: PRI
Peer Details:
_________________________________________________________________
host=192.168.2.1
disallow=all
allow=ulaw
qualify=yes
type=friend
context=from-trunk
canreinvite=yes
__________________________________________________________________

Nothing else is filled in. (user context is not used in this instance)
====================
ALSO Make sure "Allow Anonymous Inbound SIP Calls?" = YES (at least for now)
ALSO Inbound Routes: Have an Any DID + Any CID routed to a valid destination (such a test extension), that is to say in the route, leave both the DID and CID fields blank.
=====================
Where I have ulaw, you probably want to put alaw
=====================

(I go sleep now... good luck)