Alcatel Unleashed

Hi all,
I am new to the VoIP world, and I would like to know some information about using Elastix with an alcatel OmniPCX OXO, I want to connect my elastix server to an alcatel OmniPCX PBX OXO using siptrunk, my goal is to allow all the voip extensions (40) on the elastix server to connect to the phones extensions in the alcatel PBX, and allow them to make external calls using the alcatel PBX.
I would like to know if this option is possible, I also would like to know what I need to do, in the alcatel to allow this connection;
If anyone has this type of configuation working I would also apreciatte coments about the sip trunk performance.
I am using elastix 2.0.3 64 BITS.
Thank You

Nuno Martins

Yes, I have this king of connection. I even use Hyla Fax on Elastix with great results (OXO unlike OXE supportts Fax over 711).
In my config OXO detects fax on all DID's and routes faxes through SIP TG to Elastix HylaFax (you need Integrated AA license in OXO for this), and voice calls to normal extension in OXO. Voice calls are also ok w/o issues between OXO and Asterisk.
For Asterisk config refer to:
viewtopic.php?f=227&t=14231

OXO config (w/o authentication/registrar in OXE):

thank you for the reply

Hi, well I know it´s been a time since this problem was posted, but I have a question, how i can need to configurate my elastix and the PBX Alcatel, I looked my configuration in the PBX and it´s the same like you, but i can´t made the connection between my extensions from the PBX to my SIP extensions that I create in elastix, please someone can´t help me this make me crazy because I don´t know what can be, and sorry for my english but I´m learning every day

Post your traces from both asterisk and OXO.

tot3nkopf wrote:Post your traces from both asterisk and OXO.

sorry answered i little late but well here is
well this is all the configuration that I have for the moment

In my sip trunk of elastix it´s

Outgoing Settings

host=191.168.100.1
username=alis
fromuser=alis
secret=nolose
type=peer
insecure=yes
context=from-pstn
fromdomain=192.168.96.99
transfer=yes
inmediate=no
dtmfmode=outofband
dtmf=info
nat=no

Incoming Settings

context=from-pstn
host=192.168.100.1
type=user
disallow=all
allow=all
insercure=yes
allow=all
qualify=yes


SIP ABC-F:
─Review/Modify: Trunk Groups──────────────────────────────────────────────────┐
│
Node Number (reserved) : 1 │
Trunk Group ID : 7 │
│
Trunk Group Type + T2 │
Trunk Group Name : SIP ABC-F │
UTF-8 Trunk Group Name : --------------------------------------- │
Number Compatible With : -1 │
Remote Network : 31 │
Shared Trunk Group + False │
Special Services + Nothing │
Node number : 1 │
Transcom Trunk Group + False │
Auto.reserv.by Attendant + False │
Overflow trunk group No. : -1 │
Tone on seizure + False │
Private Trunk Group + False │
Q931 Signal variant + ABC-F │
SS7 Signal variant + No variant │
Number Of Digits To Send : 0 │
Channel selection type + Quantified │
Auto.DTMF dialing on outgoing call + NO │
T2 Specification + SIP │
Homogenous network for direct RTP + NO │
Public Network COS : 31 │
DID transcoding + False │
Can support UUS in SETUP + True │
│
Implicit Priority │
│
Activation mode : 0 │
Priority Level : 0 │
│
Preempter + NO │
Incoming calls Restriction COS : 10 │
Outgoing calls Restriction COS : 10 │
Callee number mpt1343 + NO │
Overlap dialing + YES │
Call diversion in ISDN + YES │
│
──────────────────────────────────────────────────────────────────────────────┘


┌─Review/Modify: Trunk Group──────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Trunk Group ID : 7 │
│ Instance (reserved) : 1 │
│ │
│ Trunk Group Type + T2 │
│ T2 Specification + SIP │
│ Public Network Ref. : ------ │
│ VG for non-existent No. + YES │
│ Entity Number : 0 │
│ Supervised by Routing + NO │
│ VPN Cost Limit for Incom.Calls : 0 │
│ Immediate Trk Listening if VPNCall + YES │
│ VPN TS % : 50 │
│ CSTA-Monitored + NO │
│ Max.% of trunks out CCD : 0 │
│ Ratio analog.to ISDN cost : ------ │
│ TS Distribution on Accesses + YES │
│ Quality profile for voice over IP + Profile #1 │
│ IP Compression Type + G 711 │
│ Use of volume in system + YES │
│ Announcement for dial tone + NO │
│ Announcement for Ring tone + NO │
│ Private to Public Overflow + YES │
│ End-to-end dialing + NO │
│ DTMF end-to-end signal. + NO │
│ Trunk group used in DISA + NO │
│ DISA Secret Code : ---- │
│ Routing To Manager + NO │
│ Trunk COS : 31 │
│ Sending of Progress message + YES │
│ No. of digits unused (ISDN) : 0 │
│ B Channel Choice + YES │
│ Channels: Attendant Control (Rsvd) : 0 │
│ Redirection For ACD (Dissuasion) + NO │
│ DTO joining + NO │
│ Consultation Call On B Channel + NO │
│ Automated Attendant + NO │
│ Calling party Rights COS : 0 │
│ TS Overflow + YES │
│ Number To Be Added : -------- │
│ Charge Calling And ADN Creation + YES │
│ Logical Channel + 1__15 & 17__31 │
│ Use Split Access + NO │
│ Heterogeneous Remote Network + NO │
│ COS Restrictions - Barring mode + Not Restricted / Not barred │
│ ARS Class of service : 31 │
│ External Access Server + NO │
│ CSTA Tracking MCDU Trk : -------- │
│ │
└─────────────────────────────────────────────────────────────────────────┘
SIP GW:

┌─Review/Modify: SIP Gateway───────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ │
│ SIP Subnetwork : 10 │
│ SIP Trunk Group : 7 │
│ IP Address : 192.168.96.99 │
│ Machine name - Host : ncsnode │
│ SIP Proxy Port Number : 5060 │
│ SIP Subscribe Min Duration : 1800 │
│ SIP Subscribe Max Duration : 86400 │
│ Session Timer : 1800 │
│ Min Session Timer : 900 │
│ Session Timer Method + RE_INVITE │
│ DNS local domain name : --------------------------------------- │
│ DNS type + DNS A │
│ SIP DNS1 IP Address : --------------------------------------- │
│ SIP DNS2 IP Address : --------------------------------------- │
│ SDP in 18x + True │
│ Cac SIP-SIP + False │
│ INFO method for remote extension + False │
│ Dynamic Payload type for DTMF : 97 │
│ │
└──────────────────────────────────────────────────────────────────────────────┘


SIP Ext GW:

┌─Review/Modify: SIP Ext Gateway───────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ SIP External Gateway ID : 0 │
│ │
│ Gateway Name : Asterisk │
│ SIP Remote domain : 192.168.96.99 │
│ PCS IP address : --------------------------------------- │
│ SIP Port Number : 5060 │
│ SIP Transport Type + UDP │
│ RFC3262 Forced use + False │
│ Belonging Domain : --------------------------------------- │
│ Registration ID : --------------------------------------- │
│ Registration ID in P_Asserted + True │
│ Registration timer : 0 │
│ SIP Outbound Proxy : --------------------------------------- │
│ Supervision timer : 0 │
│ Trunk group number : 100 or 101 │
│ Pool Number : -1 │
│ Outgoing realm : --------------------------------------- │
│ Outgoing username : --------------------------------------- │
│ │
│ Outgoing Password : ---------- │
│ Confirm : ---------- │
│ │
│ Incoming username : --------------------------------------- │
│ │
│ Incoming Password : ---------- │
│ Confirm : ---------- │
│ │
│ RFC 3325 supported by the distant + True │
│ DNS type + DNS A │
│ SIP DNS1 IP Address : --------------------------------------- │
│ SIP DNS2 IP Address : --------------------------------------- │
│ SDP in 18x + True │
│ Minimal authentication method + SIP None │
│ INFO method for remote extension + False │
│ Send only trunk group algo + False │
│ To EMS + True │
│ Dynamic Payload type for DTMF : 97 │
│ │
└──────────────────────────────────────────────────────────────────────────────┘


Routing number:


┌─Review/Modify: Prefix Plan──────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Number : 1 │
│ │
│ Prefix Meaning + Routing No. │
│ Network Number : 5 │-----Here i put 10 because that it's my network start whit 10.X.X.X
│ Node Number/ABC-F Trunk Group : 7 │
│ Number of Digits : 3 │
│ Number With Subaddress (ISDN) + NO │
│ Default X25 ID.pref. + NO │
│ │
└─────────────────────────────────────────────────────────────────────────┘

┌─Review/Modify: Network Routing Table───────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Network Number : 5 │
│ │
│ Rank of First Digit to be Sent : 1 │
│ Incoming identification prefix : -------- │
│ Protocol Type + ABC_F │
│ Numbering Plan Descriptor ID : 11 │
│ ARS Route list : 0 │
│ Schedule number : -1 │
│ ATM Address ID : -1 │
│ Network call prefix : -------- │
│ City/Town Name : -------------------- │
│ Send City/Town Name + False │
│ Associated Ext SIP gateway : 0 │
│ Enable UTF8 name sending + True │
│ │
└────────────────────────────────────────────────────
I don´t know what could be the problem, I read every day a different forum, and in all the forums said that it´s very easy but no for me, it´s my firt time that i´m doing something like this

I was saw a configuration with the same configuration like i made, only that chance in some parameters but it´s the same, my problem is that i can make a phone call, it´s said that "service unavailable", what could be the problem, and again sorry for my redaction.

I was looking i need to sincronice my sip trunk with the ISDN trunk? maybe there could be the problem, well i really hope that you can tell me what could be

hi
i see that you are configuring an OXE not an OXO?!
1-sip abcf trunk
if Number Compatible With : -1
so Number Of Digits To Send : 29 instead of 0
remote network =15
trunk publiccos =31 chek facilitie an authorisation for this cos (by default may be all is not allowed area and compatibilty trukg -trunk group)

2-sip external gateway
ip adress is same as oxe ?! it should b the asterisk one and trunk group is 7

3-SIP GW:

┌─Review/Modify: SIP Gateway───────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ │
│ SIP Subnetwork : -1 │
│ SIP Trunk Group : -1│
│ IP Address : 192.168.96.99 │

4-Network Number : 5 │-----Here i put 10 because that it's my network start whit 10.X.X.X
so
Review/Modify: Network Routing Table───────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Network Number : 5 >>>>10││ │
hope it can help

Hi again, well i changue the mistakes for the last post, i have a question, mm de remote network and network number what could be if my network it´s 10.X.X.X and the well all my networks start whit 10, that´s my question, and in my configuration I have already that´s suppost to be, and again thank U for the help, thanks

haroun wrote:hi
i see that you are configuring an OXE not an OXO?!
1-sip abcf trunk
if Number Compatible With : -1
so Number Of Digits To Send : 29 instead of 0
remote network =15
trunk publiccos =31 chek facilitie an authorisation for this cos (by default may be all is not allowed area and compatibilty trukg -trunk group)

2-sip external gateway
ip adress is same as oxe ?! it should b the asterisk one and trunk group is 7

3-SIP GW:

┌─Review/Modify: SIP Gateway───────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ │
│ SIP Subnetwork : -1 │
│ SIP Trunk Group : -1│
│ IP Address : 192.168.96.99 │

4-Network Number : 5 │-----Here i put 10 because that it's my network start whit 10.X.X.X
so
Review/Modify: Network Routing Table───────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Network Number : 5 >>>>10││ │
hope it can help

Sorry i have another question, with the configuration that you made you can comunicate also with the analog telephones?

At least did you looked at this post:
viewtopic.php?f=227&t=14231 ???
And if yes why don't you post on that topic, as you have an OXE, not an OXO?