We are new to setting up Adtrans. We are trying to use the TA924E for Analog Patient phones and have SIP trunks register directly to the Adtran and not use a PBX in the mix.
We have been able to get the trunks to work in bound routing with DIDs and we can call from station to station but on outbound calling are calls get rejected with fast busy signals.
We can keep working the issue but wondering if the fact that we do not have a PBX means it will never work outbound.
PBX is not required. You just need a SIP server to terminate the SIP messages.
Can you post your config and make sure to remove all passwords.
Also enter the following debug while trying to place a call
debug sip stack messages
debug voice verbose
debug sip cldu
debug interface fxs
-Mark
Yes, you can. If everything else is working, the fast busy on outdial is most likely an issue with the way your grouped-trunk is set up for outbound dialing. Since stations aren't a trunk, you can't to my knowledge explicitly allow them to make calls. What I do is define an ani-list, and then allow that list. For example:
voice ani-list ValidAni
ani XXX-XXX-XXXX
voice grouped-trunk OUTBOUND
trunk T01
accept 1-NXX-XXX-XXXX cost 200
accept NXX-XXX-XXXX cost 200
accept 911 cost 200
permit list ValidAni
!deny all other trunks
!deny all other ani
voice user 100
caller-id-override internal-number 212-555-1212
The voice feature-mode setting might also have implications on station-to-trunk calls. I'm using "voice feature-mode network", but the alternative is "voice feature-mode local".
Here is a config I use where I drop calls via SIP trunks from our SBC. If I understand correctly you want each port to be it's own tn on the fxs.
!
ip firewall
no ip firewall alg msn
no ip firewall alg h323
!
interface eth 0/1
ip address x.x.x.x 255.255.x.x
media-gateway ip primary
no shutdown
!
interface fxs 0/1
no shutdown
!
interface fxs 0/2
no shutdown
!
interface fxs 0/3
no shutdown
!
interface fxs 0/4
no shutdown
!
interface fxs 0/5
no shutdown
!
interface fxs 0/6
no shutdown
!
interface fxs 0/7
no shutdown
!
interface fxs 0/8
no shutdown
!
interface fxs 0/9
no shutdown
!
interface fxs 0/10
no shutdown
!
interface fxs 0/11
no shutdown
!
interface fxs 0/12
no shutdown
!
interface fxs 0/13
no shutdown
!
interface fxs 0/14
no shutdown
!
interface fxs 0/15
no shutdown
!
interface fxs 0/16
no shutdown
!
ip access-list extended sip
permit udp x.x.x.x 0.0.0.0 eq 5060 any log !siptrunk provider
deny udp any eq 5060 any log
!
no ip tftp server
no ip tftp server overwrite
ip http server
ip http secure-server
ip snmp agent
no ip ftp server
no ip scp server
no ip sntp server
voice feature-mode network
voice forward-mode network
!
!
!
!
!
!
!
voice dial-plan 1 local NXX-NXX-XXXX
voice dial-plan 2 long-distance 1-NXX-NXX-XXXX
voice dial-plan 3 international 011XXXXXXXXXXXXXXXXXX
!
voice trunk T01 type sip
description "SIP trunk provider"
match dnis "NXX-NXX-XXXX" substitute "1-NXX-NXX-XXXX"
match dnis "NXX-XXXX" substitute "1-NPA-NXX-XXXX"
sip-server primary x.x.x.x
outbound-proxy primary x.x.x.x
max-number-calls 24
default-ring-cadence internal
!
!
voice grouped-trunk MAIN
no description
trunk T01
accept $ cost 0
!
voice user NPANXX0001 !<- replace with TN
connect fxs 0/1
first-name "NPANXX0001" !<- replace with TN
password encrypted "2723ef1bace523e88dc1839aa58a7da49d2f"
no special-ring-cadences
!
voice user NPANXX0002 !<- replace with TN
connect fxs 0/2
first-name "NPANXX0002" !<- replace with TN
password encrypted "2723ef1bace523e88dc1839aa58a7da49d2f"
no special-ring-cadences
!
! repeat to 24
!
ip sip
!
no ip sip registrar authenticate
!
ip rtp quality-monitoring
ip rtp quality-monitoring sip
Juran,
When a Voice user places a call, first place the switchboard looks to send the call is a trunk. So it will send it out T01. Does your SIP Trunk require registrations for placing outbound calls? I notice you don't have any registrations configured on it.
Then the SIP server will send call back down to 900 and it will ring FXS user.
To see what is going on, I need to see debug. Enter the debug that I posted in previous post and place an outbound call. I can then tell you what the problem is.
-Mark
this maybe your issue "permit list ValidAni" it doesn't match voice user 100