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..
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
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
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
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
!
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
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.
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