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bflippen
New Contributor

How to route to a differnt SIP trunk based upon DNIS

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644

Currently 4 SIP trunks coming in/out and 4 PRI interfaces in use.

They want to add an outbound asterix dialer to the mix connected via a SIP trunk.

This is a job I inherited. Still not up to speed. If there is a good doc to read for this please advise. I am a complete newb on this.

So first a few questions about the current config below to better help me understand a few things:

1) for the Voice grouped-trunk 911, it has:

     trunk T03

     TrunkT01

Does it utilize the top down method for access? Meaning T03 first and if full/unavailable?

2) We have 4 grouped trunks

1 accepts any ($)

1 accepts only 911

1 accepts specific numbers

1 facing the PBX that accepts any number.

How is priority set on these? Does the unit look through each of the grouped trunks for a specific match then routes out accordingly? Or is there a mechanism I am not seeing?

3) what is !deny ?  

Since I don't want people to do  my job entirely, I was thinking that all I have to do is: If I am wring then could someone do my job entirely?

*****************************************

Add

trunk 06 to PBX_Trunks

voice trunk T06 type sip

  description "Dialer"

  sip-server primary 192.168.1.1 (The local IP address of the asterix dialer)

  codec-list SIPCodecs both

Voice group-trunk SIP-Asterix

     Trunk T06

     accept 5555551111 cost 0

     permit list SIP_Trunks

     !deny all other trunks

     !deny all other ani

********************************************

Here is the current config:

interface t1 0/1

  description PRI1

  clock source line

  timing-domain 1

  system-timing primary

  fdl att

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

interface t1 0/2

  description PRI2

  clock source line

  timing-domain 1

  fdl att

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

interface t1 0/3

  description PRI3

  clock source line

  timing-domain 1

  fdl att

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

interface t1 0/4

  description PRI4

  clock source line

  timing-domain 1

  fdl att

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

!

interface pri 1

  isdn name-delivery display

  connect t1 0/1 tdm-group 1

  role network b-channel-restarts enable

  no shutdown

!

interface pri 2

  isdn name-delivery display

  connect t1 0/2 tdm-group 1

  role network b-channel-restarts enable

  no shutdown

!

interface pri 3

  isdn name-delivery display

  connect t1 0/3 tdm-group 1

  role network b-channel-restarts enable

  no shutdown

!

interface pri 4

  isdn name-delivery display

  connect t1 0/4 tdm-group 1

  role network b-channel-restarts enable

  no shutdown

!

!

isdn-group 1

  connect pri 1

  connect pri 2

  connect pri 3

  connect pri 4

!

voice codec-list SIPCodecs

  codec g729

  codec g711ulaw

!

voice codec-list G711Codec

  codec g711ulaw

!

!

voice trunk-list PBX_Trunks

  trunk T05

!

voice trunk-list SIP_Trunks

  trunk T01

  trunk T02

  trunk T03

  trunk T04

!

!

voice trunk T01 type sip

  description "Local TG with list of codecs"

  sip-server primary 67.14.XXX.XXX

  codec-list SIPCodecs both

!

voice trunk T02 type sip

  description "Locall TG with only G711 codec"

  sip-server primary 67.14.XXX.XXX

  codec-list G711Codec both

!

voice trunk T03 type sip

  description "USE TG with list of codecs"

  sip-server primary 67.14.XXX.YYY

  codec-list SIPCodecs both

!

voice trunk T04 type sip

  description "USE TG with only G711 codec"

  sip-server primary 67.14.XXX.YYY

  codec-list G711Codec both

!

voice trunk T05 type isdn

  description "PRIs"

  resource-selection linear ascending

  connect isdn-group 1

  no early-cut-through

  rtp delay-mode adaptive

!

voice grouped-trunk SIP-PBX

  trunk T03

  trunk T01

  accept $ cost 0

  permit list PBX_Trunks

  !deny all other trunks

  !deny all other ani

!

voice grouped-trunk 911

  trunk T02

  trunk T04

  accept 911 cost 0

  permit list PBX_Trunks

  !deny all other trunks

  !deny all other ani

!

!

voice grouped-trunk MTG

  trunk T04

  trunk T02

  accept 2135551212 cost 0

  accept 4155551212 cost 0

  accept 3125551212 cost 0

  accept 2675551212 cost 0

  accept 6195551212 cost 0

  permit list PBX_Trunks

  !deny all other trunks

  !deny all other ani

!

!

voice grouped-trunk PBX-PRIs

  trunk T05

  accept $ cost 0

  permit list SIP_Trunks

  !deny all other trunks

  !deny all other ani

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jayh
Honored Contributor
Honored Contributor

Re: How to route to a differnt SIP trunk based upon DNIS

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First let's clarify terminology - DNIS is the number dialed, ANI is the calling number.

The default behavior of the unit is to route based on the DNIS, dialed number.

Most specific route wins, meaning if one grouped-trunk has "accept $" and another has "accept 213XXXXXXX" then all calls to numbers in the 213 area code will go out the second trunk. If a third grouped-trunk has "accept 2135551212" then calls to that specific number will go there.

Your configuration has an added feature called SABR  for "Source ANI-Based Routing" where the origin of the call is also used to make routing decisions. This is where the trunks are defined into trunk-lists and each grouped-trunk specification matches both the destination DNIS and the source trunk-list.  The "!deny all other trunks" and "!deny all other ANI" are inserted automatically as comments to let you know that only the permitted trunk-list or ani-lists will use that route.

Your example has "permit list SIP_Trunks" for the Asterisk. If your intent is to route calls originating from SIP to the Asterisk, you should be good to go other than possible registration/authentication between the Adtran and Asterisk to allow it to accept the calls.

If your intent is to have the PBX calls flow to the Asterisk, then substitute "permit list PBX_Trunks" in the configuration for grouped-trunk SIP-Asterix.

If you want to route all calls to 5555551111 to the Asterisk regardless of origin, then don't specify a trunk-list at all for that grouped-trunk.

View solution in original post

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2 Replies
jayh
Honored Contributor
Honored Contributor

Re: How to route to a differnt SIP trunk based upon DNIS

Jump to solution

First let's clarify terminology - DNIS is the number dialed, ANI is the calling number.

The default behavior of the unit is to route based on the DNIS, dialed number.

Most specific route wins, meaning if one grouped-trunk has "accept $" and another has "accept 213XXXXXXX" then all calls to numbers in the 213 area code will go out the second trunk. If a third grouped-trunk has "accept 2135551212" then calls to that specific number will go there.

Your configuration has an added feature called SABR  for "Source ANI-Based Routing" where the origin of the call is also used to make routing decisions. This is where the trunks are defined into trunk-lists and each grouped-trunk specification matches both the destination DNIS and the source trunk-list.  The "!deny all other trunks" and "!deny all other ANI" are inserted automatically as comments to let you know that only the permitted trunk-list or ani-lists will use that route.

Your example has "permit list SIP_Trunks" for the Asterisk. If your intent is to route calls originating from SIP to the Asterisk, you should be good to go other than possible registration/authentication between the Adtran and Asterisk to allow it to accept the calls.

If your intent is to have the PBX calls flow to the Asterisk, then substitute "permit list PBX_Trunks" in the configuration for grouped-trunk SIP-Asterix.

If you want to route all calls to 5555551111 to the Asterisk regardless of origin, then don't specify a trunk-list at all for that grouped-trunk.

View solution in original post

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bflippen
New Contributor

Re: How to route to a differnt SIP trunk based upon DNIS

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the config was existing SIP trunks connected to the PBX.They are planning on adding an additional dialer connected by SIP.

The dialer was going to dial out on the existing SIP trunks and if a particular number was called inbound (DNIS) it would route to the DIaler instead of the PBX. Thanks for the SABR explanation as well.

The clarification on the specific match confirmed what I was thinking.

also thanks for the !deny explanation.