Alcatel Unleashed

Hello all,

At our company we have an omnipcx enterprise with sip licenses activated. We now want to create a queue test with an Elastix sever as a SIP client. Dialing out is working like a charm, but incomming rings are lost (checked with tcpdump, it is not arriving at the Elastix server).
My question is now, has anyone any experience with this, and if so, what are the correct setings in the incomming settings in the trunk (user context, user details and registration string) to get this working? One of the network cards is sconnected directly to the same network as the PBX.

Kind regards,
Roland.

Post your configuration on both OXE and Elastix.

tot3nkopf wrote:Post your configuration on both OXE and Elastix.

Configuration was pretty straight forward, the clou was that the registration timeout had to be increased globaly to at least 3600.

Thanks all,

Roland

I use Elastix in my lab. I have successfully connected OXE and Asterisk over both H.323 (ooh on asterisk) and SIP trunks. On SIP trunk I have made some tests and only succeeded without authentication (I have either Bad auth or Unauthorised from Asterisk side when trying).
Without external gateway registration I have full services: name display, transfer, tranzit, DTMF, etc
Can you post the configs from both OXE and Elastix?

Can you post the conf to do this?

users 2XX OXE <--> 1XX Asterisk (Elastix)


Elastix config (pasted from web interface -->if needed I can provide .conf files):

peer config:
host=192.168.0.101
username=asterisk
fromuser=asterisk
secret=asterisk
type=peer
insecure=very
context=from-pstn
fromdomain=192.168.0.105
transfer=yes
immediate=no
dtmfmode=outofband
dtmf=rfc2833
nat=no
canreinvite=no ;yes
qualify=no; yes





OXE:

-------------------------------------------------------------------
Numb | Name | Type | Var. | Node | Pfx
-------------------------------------------------------------------
TG 1 | Orange | T0 | ISDN | 1 => local | ** or *#*#01
TG 10 | H.323 | T2-IP | ISDN | 1 => local | No pfx
TG 100 | SIP ABC-F | T2-SIP | IPNS | 1 => local | No pfx
TG 101 | SIP ISDN | T2-SIP | ISDN | 1 => local | No pfx
-------------------------------------------------------------------
LK 2000 | Loop | T2HYBRID | IPNS | to network_node 0-1
-------------------------------------------------------------------


H.323

─Review/Modify: Trunk Groups──────────────────────────────────────────────────┐
│
Node Number (reserved) : 1 │
Trunk Group ID : 10 │
│
Trunk Group Type + T2 │
Trunk Group Name : H.323 │
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 + ISDN all countries │
SS7 Signal variant + No variant │
Number Of Digits To Send : 0 │
Channel selection type + Quantified │
Auto.DTMF dialing on outgoing call + NO │
T2 Specification + IP │
Homogenous network for direct RTP + NO │
Public Network COS : 0 │
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 + NO │
│
──────────────────────────────────────────────────────────────────────────────┘
─Review/Modify: Trunk Group──────────────────────────────────────────────┐
│
Node Number (reserved) : 1 │
Trunk Group ID : 10 │
Instance (reserved) : 1 │
│
Trunk Group Type + T2 │
T2 Specification + IP │
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 % : 100 │
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 : 19 │
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 + NO │
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 ABC-F:
─Review/Modify: Trunk Groups──────────────────────────────────────────────────┐
│
Node Number (reserved) : 1 │
Trunk Group ID : 100 │
│
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 : 100 │
│ 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 ISDN:

┌─Review/Modify: Trunk Groups──────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Trunk Group ID : 101 │
│ │
│ Trunk Group Type + T2 │
│ Trunk Group Name : SIP ISDN │
│ 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 + ISDN all countries │
│ 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 : 101 │
│ 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 : 100 │
│ IP Address : 192.168.0.101 │
│ 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 : 192.168.0.105 │
│ 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 │
│ Node Number/ABC-F Trunk Group : 100 │
│ 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 │
│ │
└────────────────────────────────────────────────────────────────────┘


ARS:

┌─Review/Modify: Prefix Plan──────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Number : 0 │
│ │
│ Prefix Meaning + ARS Prof.Trg Grp Seiz.with overlap │
│ Discriminator No. : 0 │
│ │
└─────────────────────────────────────────────────────────────────────────┘

┌─Review/Modify: Discriminator Rule────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Discriminator No. : 0 │
│ Call Number : 1 │
│ │
│ Area Number : 1 │
│ ARS Route List Number : 2 │
│ Schedule Number : -1 │
│ Number of Digits : 3 │
│ │
└──────────────────────────────────────────────────────────────────────────────┘

┌─Review/Modify: ARS Route───────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ ARS Route list : 2 │
│ Route : 1 │
│ │
│ Name : Asterisk │
│ Trunk Group Source + Route │
│ Trunk Group : 101 │
│ No.Digits To Be Removed : 0 │
│ Digits To Add : ------------------------------ │
│ Numbering Command Tabl. ID : 1 │
│ VPN Cost Limit : 0 │
│ Protocol Type + Dependant on Trunk Group Type │
│ NPD identifier : 255 │
│ Route Type + Public │
│ ATM Address ID : -1 │
│ Preempter + False │
│ │
│ Quality │
│ │
│ [ Add ] [ Remove ] [ Next ] [Previous] │
│ │
│ Quality + Speech │
│ │
└────────────────────────────────────────────────────────────────────────────┘

┌─Review/Modify: Numbering Command Table───────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Table ID : 1 │
│ │
│ Carrier Reference : 0 │
│ Command : --------------------------------------- │
│ Associated Ext SIP gateway : 0 │
│ │
└──────────────────────────────────────────────────────────────────────────────┘

Thanks.

Hi can you reattached your image files,

I'm really confuse to connect the OXE with Elastix,
I'm do almost same at all of your setup
From Elastix SIP Phone I'm using Dial #1 to connect to Alcatel, it's successful, After #1 I can use both call to Extension or use Trunk to call out
Example #1 <1001> <<< call to extension 1001
#1 9(PrefixPlan) 856xxxxx <<< call to landline number using the trunk
When call using the trunk, sometimes have delayed or problem for connected.
My problems is I can't call from Alcatel to Elastix
I'm planing to use #1 for connecting to Elastix, so I can call to Extension at Elastix or Use the SIP trunk on Elastix.

thanks for helping

Check again.

Hi thanks for reply and reuploading again

my I know what is the function of H323 trunk?

I'm don't understand on this string ooh323/$OUTNUM$@192.168.0.123

what is 192.168.0.123 ip?

Right now my configuration is using SIP trunk and almost similar with ur setup already..
Alcatel IP = 192.168.100.1 and Elastix IP 192.168.100.101
My connection from Elastix to Alcatel is no problem, only sometimes will delay, mostly when I tried to use trunk lin on Alcatel. For call to Alcatel Extension it's pretty good.
My connection from Alcatel to Elastix can't connect, having a problem.. I'm confuse how to make route from Alcatel to Elastix.. already tried the method from you

Sorry I'm newbie on Alcatel , just try to improve my skills. I'm very thanks for you answer and if you would like to explain more detail to me how to make route from Alcatel to Elastix.
Thanks