Multiple SIP TRUNK on 900 series

[apenichet]

Guys,

I am currently trying to get 2 sip trunks to work at the same time on a 916E. We currently have a Metaswitch with a Permitta SBC. Here is the current config....

voice trunk T01 type sip

  description "Metaswitch"

  sip-server primary 68.68.XXX.138

  outbound-proxy primary iad.myt3voip.com

  authentication username "adtran" password encrypted XXXXX

  max-number-calls 60

  codec-group Fax

voice trunk T05 type sip

  description "Metaswitch911"

  sip-server primary 68.68.XXX.138

  outbound-proxy primary iad.myt3voip.com

  authentication username "adtran911" password encrypted XXXXX

  max-number-calls 24

  codec-group Trunk

As you can see they both have the same ip address. Well looks like my switch does not like this. It was recommended to have the T05 trunk on a different ip address. So then I tried the following. And used a different ip address.

voice trunk T05 type sip

  description "Metaswitch911"

  sip-server primary 68.68.XXX.139

  outbound-proxy primary iad.myt3voip.com

  authentication username "adtran911" password encrypted XXXXX

  max-number-calls 24

  codec-group Trunk

As soon as I set this up my SIP TRUNK in the switch it goes to alarm. It looks like the T05 is still showing the original ip address with .138 which is causing the failure. Is there anything special that needs to be done when setting up multiple SIP TRUNK to make sure they differentiate them selves..

[Anonymous]

I had the same issue when using the same IP address on multiple SIP trunks. Also on Meta. This is how my setup looks to resolve the issue.

!

voice ani-list MatchAll

  ani $

!

voice trunk-list SIP

  trunk T02

  trunk T11

  !

!

voice grouped-trunk SIP_TO_E911

  description "E911 Services"

  trunk T02

  accept 911 cost 0

  accept 9911 cost 0

  reject M

!

!

voice grouped-trunk "SIP TO GDS"

  description "GDS SIP Trunk"

  trunk T11

  accept NXX-NXX-XXXX cost 0

  accept 1-NXX-NXX-XXXX cost 0

  accept 1-800-NXX-XXXX cost 0

  accept 1-888-NXX-XXXX cost 0

  accept 1-877-NXX-XXXX cost 0

  accept 1-866-NXX-XXXX cost 0

  accept 1-855-NXX-XXXX cost 0

  accept 011-$ cost 0

  accept 0-NXX-NXX-XXXX cost 0

  accept NXX-XXXX cost 0

  accept 337-NXX-XXXX cost 0

  reject NXX-976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

  reject 976-XXXX

  deny list SIP

  permit list MatchAll

  !deny all other trunks

  !deny all other ani

!

  end

Corected my issue. Hope it helps.

ClintB

[ftwrobert]

We too have a Metaswitch, However, we are not using the permitta SBC (we are using kamailio). I had the same problem when setting up multiple voice trunk's.

My solution is to have a user/pass on the voice trunk (which the PRI will use) and a user/pass on each of the voice users.

BendTel-Test-9xx#show run

Building configuration...

!

!

! ADTRAN, Inc. OS version R10.9.1.E

! Boot ROM version 14.05.00.SA

! Platform: Total Access 908e (2nd Gen), part number 4242908L1

ip default-gateway 66.39.xxx.xxx

!

domain-name "bendtel.net"

name-server 66.39.xxx.xxx 66.39.xxx.xxx

!      

qos map VOIP 10

  match dscp ef

  priority percent 75

  set precedence 5

!

interface eth 0/1

  ip address  66.39.xxx.xxx  255.255.255.xxx

  media-gateway ip primary

  qos-policy out VOIP

  no shutdown

!

interface t1 0/3

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

interface pri 1

  isdn name-delivery setup

  connect t1 0/3 tdm-group 1

  digits-transferred 4

  no shutdown

!

interface fxs 0/1

  impedance 600r

  no shutdown

!

isdn-group 1

  connect pri 1

!

sip

sip udp 5060

no sip tcp

!

voice feature-mode network

voice forward-mode network

voice conferencing-mode local

!

voice spre-mode local

!

voice dial-plan 1 user1 [0,1]N11

voice dial-plan 2 user1 N11

voice dial-plan 3 user1 NXXNXXXXXX

voice dial-plan 4 user1 1NXXNXXXXXX

voice dial-plan 5 user1 011

voice dial-plan 6 user1 00

voice dial-plan 7 user1 555XXXX

voice dial-plan 8 user1 10[0,2-9]

voice dial-plan 9 user1 200

voice dial-plan 10 user1 11[0,1,4-9]X

voice dial-plan 11 user1 11[2,3]XX

voice dial-plan 12 user1 101XXXX

voice dial-plan 13 user1 95[0,8,9]XXXX

!

voice codec-list G711u

  codec g711ulaw

!

voice trunk T00 type sip

  description "SBC1 - SIP to PRI/POTS"

  sip-server primary 66.39.xxx.xxx

  registrar primary 66.39.xxx.xxx

  register 5413234055 auth-name "5413234055" password encrypted "19103e60512348abdeb49ab254d8f6fb7ecb"

  codec-list G711u both

  authentication username "5413234055" password encrypted "3138b28f6385e193f11548832c6893169202"

!

voice trunk T01 type isdn

  description "PBX - SIP to PRI"

  resource-selection circular descending

  connect isdn-group 1

  early-cut-through

  rtp delay-mode adaptive

!

voice grouped-trunk SIP

  trunk T00

  accept $ cost 0

!

voice grouped-trunk PRI

  trunk T01

  accept $ cost 0

!

voice user 5413234048

  connect fxs 0/1

  password encrypted "24209cc96b444d215061178c14ab5e607014"

  did "5413234048"

  sip-identity 5413234048 T00 register auth-name "5413234048" password encrypted "19103e60512348abdeb49ab254d8f6fb7ecb"

  sip-authentication password encrypted "3138b28f6385e193f11548832c6893169202"

!

sip qos dscp 24

!

[Anonymous]

Apenichet,

I just thought I would check in with you to see if you had resolved the issue.  Did one of the suggestions above help you?  If so, you may want to mark the answer as "Correct" or "Helpful".  If not, we may need to know more about why the second SIP trunk is needed in order to help solve the problem.

Thanks!

David

[apenichet]

I was able to get the call routing worked perfectly. However I had to use a different SBC in my network in order to get the 2nd SIP TRUNK to work correctly. It looks like the IAD is not able to use the secondary ip address that I assigned to the same interface and to the sip trunk. The SIP messages where still coming in with the original ip address. I even called support and they stated that the IAD cannot do that.

[Anonymous · Accepted solution]

Apenichet,

It sounds like you have found a solution that will work for you then.  To summarize, the source of the SIP and RTP is controlled by the media-gateway command on the egress interface, and there is only one media-gateway IP address for a given interface.  In order to use different source addresses for SIP traffic sourced from the Adtran unit itself, you would need separate interfaces.  These could be separate physical interfaces, ethernet subinterfaces, frame-relay subinterfaces, or even something like a GRE tunnel interface. 

Thanks!

David