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New Contributor

Can we use a TA924E 2nd Gen with SIP trunks to Analog stations with NO PBX

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.

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Valued Contributor II
Valued Contributor II

Re: Can we use a TA924E 2nd Gen with SIP trunks to Analog stations with NO PBX

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

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New Contributor

Re: Can we use a TA924E 2nd Gen with SIP trunks to Analog stations with NO PBX

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".

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New Contributor

Re: Can we use a TA924E 2nd Gen with SIP trunks to Analog stations with NO PBX

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

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Valued Contributor II
Valued Contributor II

Re: Can we use a TA924E 2nd Gen with SIP trunks to Analog stations with NO PBX

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

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Contributor
Contributor

Re: Can we use a TA924E 2nd Gen with SIP trunks to Analog stations with NO PBX

this maybe your issue "permit list ValidAni"  it doesn't match voice user 100