Dial Plan

[npatel]

1. Brekeke Product Name and version:Brekeke SIP Server , Version 2.4.5.5 Evaluation

2. Java version:

3. OS type and the version:Microsoft Windows Server 2003 SP2

4. UA (phone), gateway or other hardware/software involved:X-lite version 3 build 56125

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :

6. Your problem:
Hello All,
Sorry for SPAM. I have just installed the Free evaluation of SIP server.
without any dial plan I am able to MAKE/Recieve calls from EACH Registered X-LITE soft phones on two different computers.

I can't understand any thing on DIAL Plan.
Can anyone please let me know the screen shot for working dial plan on Brekeke sip server?

I have Avaya PBX and ISDN lines.

I want to test SIP BREKEKE SIP server end points to make/receive calls through AVAYA PBX.
How should I configure my DIAL plan on BREKE Sip Server?
How should I configure the SIP trunk between PBX and BREKEKE server?

Please include the screen shot if possible of BREKEKE Sip server dial plan or email me on patel77@gmail.com

Thanks and Regards,
N Patel

[hope]

at pbx, configure to send call to brekeke ip address
at brekeke, set the following dial plans to accept/send call to pbx

Rule 1: Accept calls and send to registered user

Matching Patterns

$request = ^INVITE $addr = pbx_IP_address To = sip:(.*)@

Deploy Patterns

$auth = false To = sip:%1@

with this rule, the calls from pbx address will be accepted by Brekeke SIP Server.


Rule 2: Send calls to pbx

Matching Patterns

$request = ^INVITE To = sip:0(.+)@

Deploy Patterns

To = sip:%1@pbx_IP_address

With this rule, the calls with prefix 0 in dialing numbers will be sent to pbx IP address without prefix 0.
For example, if you dial 01113333, the call will be sent to gateway as 1113333 with Rule 2.

There are more dial plans at http://wiki.brekeke.com/wiki/Dial-Plan
or dial plan tutorial http://www.brekeke.com/download/download_sip_doc_en.php

[npatel]

Hi,

Thanks for reply. I have added the dial plan accordingly.

On PBX side - I am getting "out-of-service" for the channels configured for SIP trunk between Avaya PBX (CM 5.2) and Brekeke SIP server.

The Signalling group shows as "inservice/idel" on PBX side, configured as "SIP/TLS" having listen port on 5061 for both the side.

Is there any suggestion to make it working?

Thanks and Regards,
npatel

[hope]

[npatel]

Hi Hope,

Thank you again. I have changed the settings as per your instructions on PBX SIP TRUNK on TCP for transport mode.

Still not able to see the trunks in "inservice/idel" mode on PBX.
I have attached the settings for your reference.

I have also added the dial plan on SIP as per your email.

**********AVAYA TRUNK GROUP PROGRAMMING START*********

isplay trunk-group 101 Page 1 of 21
TRUNK GROUP

Group Number: 101 Group Type: sip CDR Reports: y
Group Name: OUTSIDE CALL COR: 1 TN: 1 TAC: 186
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: public-ntwrk Auth Code? n

Signaling Group: 101
Number of Members: 5

isplay trunk-group 101 Page 2 of 21
Group Type: sip

TRUNK PARAMETERS

Unicode Name: yes

Redirect On OPTIM Failure: 5000

SCCAN? n Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 64800

RUNK FEATURES
ACA Assignment? n Measured: both
Maintenance Tests? y



Numbering Format: public
UUI Treatment: service-provider

Replace Restricted Numbers? n
Replace Unavailable Numbers? n







Show ANSWERED BY on Display? y
PROTOCOL VARIATIONS

Mark Users as Phone? n
Prepend '+' to Calling Number? n
Send Transferring Party Information? n
Network Call Redirection? n
Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type: 127

TRUNK GROUP
Administered Members (min/max): 1/5
GROUP MEMBER ASSIGNMENTS Total Administered Members: 5

Port Name
1: T00416 OUTSIDE CA
2: T00417 OUTSIDE CA
3: T00418 OUTSIDE CA
4: T00419 OUTSIDE CA
5: T00420 OUTSIDE CA
6:
***************END*************************
*********Avaya SIGNALING GROUP START*********
SIGNALING GROUP

Group Number: 101 Group Type: sip
Transport Method: tcp
IMS Enabled? n

Near-end Node Name: clan_uk3 Far-end Node Name: npatel-srv
Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 100
Far-end Domain: eur.npatel.com

Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? n Direct IP-IP Early Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6

****************************END********************
*************Avaya SIG GROUP Status START************
status signaling-group 101
STATUS SIGNALING GROUP

Group ID: 101 Active NCA-TSC Count: 0
Group Type: sip Active CA-TSC Count: 0
Signaling Type: facility associated signaling
Group State: in-service

************************END*****************
**********Avaya Trunk group status START *******
status trunk 101

TRUNK GROUP STATUS

Member Port Service State Mtce Connected Ports
Busy

0101/001 T00416 out-of-service-NE no
0101/002 T00417 out-of-service-NE no
0101/003 T00418 out-of-service-NE no
0101/004 T00419 out-of-service-NE no
0101/005 T00420 out-of-service-NE no
**************END**********************
**************SIP BREKE Dial Plan - START*******

RULE-1

Matching Patterns

$request = ^INVITE $addr = 10.2.148.8 To = sip:(.*)@

Deploy Patterns

$auth = false To = sip:1%@

RULE-2

Matching Patterns

$request = ^INVITE To = sip:0(.+)@

Deploy Patterns

To = sip:1%@10.2.248.8 $transport = tcp

****************END****************

IS THERE ANY THING I AM MISSING?

Thanks,
NPatel

[james]

what is the PBX's IP address?
is it 10.2.148.8? or is it 10.2.248.8?

can you see a SIP packet from your PBX over TCP if you use a packet capturing tool such as Wireshark?

[npatel]

Hi,

The ip address is 10.2.248.8

I have corrected that on SIP BREKEKE Server.

On Avaya Side, I am using ARS to route the calls to SIP BREKEKE Server.
The problem is, as the Channels in that trunk group is out-of-service, its not allowing me to dial out.

Below is the trace to dial SIP BREKEKE user 4466 from Avaya PBX station 4460.

list trace tac 186 Page 1

LIST TRACE

time data

10:10:31 term trunk-group 101 cid 0x3454
10:10:31 route-pattern 102 preference 1 unavailable cid 0x3454
10:10:31 dial 4466 route:UDP|AAR
10:10:31 denial event 1012: No circ/chan avail D1=0x8916 D2=0x3454
10:10:31 dial 4466 route:UDP|AAR
10:10:31 reorder trunk-group 101 cid 0x3454
10:10:32 idle station 4460 cid 0x3454

The channels are out of service, so not forwarding any traffic on that link.
How we can make it inservice/idel?

Thanks and Regards,
npatel

[hope]

you need check with Avaya for the channels problem

[james]

it seems you need to contact Avaya's technical support to fix PBX's issue..

Can you see any SIP packet come from Avaya PBX??

[npatel]

Hello,

Thanks for your reply. I am continue to working on Avaya side through my business partner.

I understand that, Avaya CM do not support direct SIP through UDP, so we need to have AVAYA SES server in between any other SIP trunk provider.

I will configure the SES server very soon on my side and let you know if I found any problem to configure the SIP BREKE with Avaya SES server.

Thanks,
npatel

[james]

>> I understand that, Avaya CM do not support direct SIP through
>> UDP, so we need to have AVAYA SES server in between any other
>> SIP trunk provider.

Also, you can use TCP instead of UDP for SIP.


>> I will configure the SES server very soon on my side and let
>> you know if I found any problem to configure the SIP BREKE
>> with Avaya SES server.

Good luck

[npatel]

Hi,

I have configured TCP SIP trunk between the Avaya CM and BREKEKE Sip server.

I am now able to make outbound call from BREKEKE SIP Server end points.

I am not able to receive any calls from Avaya PBX.
The Avaya pbx initiates the SIP trunk channel and forwards the digits, but rejected by saying DENIAL Event 1166, which is for unregistered stations.

Any suggestion to receive calls.

My dialling plan on BREKEKE SIP Server is as follow for inbound traffic: ( RULE1)

Matchin Pattern:
$request=^INVITE
$addr=10.2.248.8
To=sip:(.*)@

Deploy Patterns

$auth = false To = sip:1%@

The working outbound traffic on BREKEKE SIP Server (RULE-2):

Matching Patterns

$request = ^INVITE To = sip:0(.+)@

Deploy Patterns

To = sip:%1@10.2.248.8 $transport = tcp

The trunk group is now up and running as given below on Avaya PBX:

status trunk 101

TRUNK GROUP STATUS

Member Port Service State Mtce Connected Ports
Busy

0101/001 T00416 in-service/idle no
0101/002 T00417 in-service/idle no
0101/003 T00418 in-service/idle no
0101/004 T00419 in-service/idle no
0101/005 T00420 in-service/idle no

The trace on PBX is as given below for dialling extension 4466(BREKEKE SIP END POINT) from extension 4460(Avaya end):

time data

11:58:40 G711A ss:off ps:20
rgn:1 [10.2.133.106]:2960
rgn:1 [10.2.248.12]:3140
11:58:41 dial 4466 route:UDP|AAR
11:58:41 term trunk-group 101 cid 0xa6a
11:58:41 dial 4466 route:UDP|AAR
11:58:41 route-pattern 102 preference 1 cid 0xa6a
11:58:41 seize trunk-group 101 member 3 cid 0xa6a
11:58:41 Setup digits 4466
11:58:41 Calling Number & Name NO-CPNumber 4460
11:58:41 Proceed trunk-group 101 member 3 cid 0xa6a
11:58:41 denial event 1166: Unassigned number D1=0x8916 D2=0x201
11:58:41 idle trunk-group 101 member 3 cid 0xa6a
11:58:46 idle station 4460 cid 0xa6a
11:58:51 TRACE COMPLETE station 4460 cid 0x0

**
If you see, the Avaya PBX forward digits to Trunk group 101 and rejected by denail event 1166 due to un-registered extension.

The 4466 is registered client on BREKEKE SIP server and I am able to dial any extension of Avaya's PBX as well as external numbers.

Any suggestion welcome.

Thanks,
N Patel

[npatel]

Hi,

I have done some modification in Rule 1 and now I am able to Receive Calls from Avaya PBX as well.

So now I have completed the integration between Avaya and BREKEKE SIP Server.

Thanks to you guys for your guidance.

The RULE 1 for incoming is as given below:

Matching Patterns

$request = ^INVITE To = sip:(.*)@

Deploy Patterns

To = %1@Ip address of SIP server $auth = off

All is well now.

Thanks,
N Patel

[hope]

can you still make call to pbx with rule-2?
maybe you need to put the Rule-2 above Rule-1.
with the modified Rule-1 on top, all calls will be routed to sip server users no matter where the call from and do not check authentication infor.

[james]

I know you did "To=sip:1%@" at first.. But it should be "To=sip:%1@" or "To=sip:%1@Ip address of SIP server".

Could you share your Avaya SIP Trunk settings?

[npatel]

HI Hope & James,

Thanks again for guiding me to make Rule 1 and Rule 2.

The Rules are as given below which are working in both directions:

Rule 1

Matching Patterns

$request = ^INVITE To = sip:0(.+)@

Deploy Patterns

To = sip:%1@10.2.248.8 $transport = tcp

Rule -2

Matching Patterns

$request = ^INVITE To = sip:(.*)@

Depoy Patterns

To=%1@10.2.133.11
$auth=off


On Avaya Side, You need to create trunk group, Signalling group as givne bleow:
**************************TRUNK - ~START**************
Trunk group details:
display trunk-group 101 Page 1 of 21
TRUNK GROUP

Group Number: 101 Group Type: sip CDR Reports: y
Group Name: BRKEKE SIP COR: 1 TN: 1 TAC: 186
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: public-ntwrk Auth Code? n

Signaling Group: 101
Number of Members: 5

display trunk-group 101 Page 2 of 21
Group Type: sip

TRUNK PARAMETERS

Unicode Name: yes

Redirect On OPTIM Failure: 5000

SCCAN? n Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 64800

display trunk-group 101 Page 3 of 21
TRUNK FEATURES
ACA Assignment? n Measured: both
Maintenance Tests? y



Numbering Format: public
UUI Treatment: service-provider

Replace Restricted Numbers? n
Replace Unavailable Numbers? n







Show ANSWERED BY on Display? y

display trunk-group 101 Page 4 of 21
PROTOCOL VARIATIONS

Mark Users as Phone? n
Prepend '+' to Calling Number? n
Send Transferring Party Information? n
Network Call Redirection? n
Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type: 127


display trunk-group 101 Page 5 of 21
TRUNK GROUP
Administered Members (min/max): 1/5
GROUP MEMBER ASSIGNMENTS Total Administered Members: 5

Port Name
1: T00416 BRKEKE SIP
2: T00417 BRKEKE SIP
3: T00418 BRKEKE SIP
4: T00419 BRKEKE SIP
5: T00420 BRKEKE SIP
6:
7:
***************END***************************

************Signalling group START********************
The SIGNALLING group details are as given below:

display signaling-group 101
SIGNALING GROUP

Group Number: 101 Group Type: sip
Transport Method: tcp
IMS Enabled? n





Near-end Node Name: clan_uk3 Far-end Node Name: npatel-srv
Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 100
Far-end Domain: eur.npatel.com

Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? n Direct IP-IP Early Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6

***************end************

*******Avaya IP NETWORK REGION START************

display ip-network-region 100 Page 1 of 19
IP NETWORK REGION
Region: 100
Location: 1 Authoritative Domain: eur.npatel.com
Name: SIP BREKEKE
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 6 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? n
UDP Port Max: 65531
DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y
Call Control PHB Value: 26 RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46 Use Default Server Parameters? y
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 3
Audio 802.1p Priority: 5
Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5

IP NETWORK REGION

INTER-GATEWAY ALTERNATE ROUTING / DIAL PLAN TRANSPARENCY
Incoming LDN Extension:
Conversion To Full Public Number - Delete: Insert:
Maximum Number of Trunks to Use for IGAR:
Dial Plan Transparency in Survivable Mode? n

BACKUP SERVERS(IN PRIORITY ORDER) H.323 SECURITY PROFILES
1 1 challenge
2 2
3 3
4 4
5
6 Allow SIP URI Conversion? y

TCP SIGNALING LINK ESTABLISHMENT FOR AVAYA H.323 ENDPOINTS
Near End Establishes TCP Signaling Socket? y
Near End TCP Port Min: 61440
Near End TCP Port Max: 61444

display ip-network-region 100 Page 3 of 19

Source Region: 100 Inter Network Region Connection Management I M
G A e
dst codec direct WAN-BW-limits Video Intervening Dyn A G a
rgn set WAN Units Total Norm Prio Shr Regions CAC R L s
1 1 y NoLimit n
2 1 y NoLimit n
3 1 y NoLimit n

display ip-network-region 100 Page 9 of 19

Source Region: 100 Inter Network Region Connection Management I M
G A e
dst codec direct WAN-BW-limits Video Intervening Dyn A G a
rgn set WAN Units Total Norm Prio Shr Regions CAC R L s
91
92
93
94
95
96
97
98
99
100 6 all
101 6 y NoLimit n all
102
103
104
105

************END*****************

With the above settings, one can connect the SIP BREKEKE SERVER with AVAYA Communication Manager.

Thanks and Regards,
npatel

[james]

Hi npatel

Thank you for sharing the information!
The information will help someone who wants to use Avaya and Brekeke.