FXS config on 904

[normrr]

We have been using the 904 2nd gen to provide SIP to PRI for customer hand-off, We are now looking at providing analog lines through the 4 FXS ports. The SIP trunks are provide by a Netsapiens switch

through ethernet, I have modified the script to route an analog line to the 904, inbound works great but when I off hook I access the PRI T1 towards the PBX. Any advice would be helpfull.

[Anonymous]

See this document on SABR.

https://supportforums.adtran.com/servlet/JiveServlet/previewBody/1862-102-2-1978/Source%20and%20ANI%20Based%20Routing%20in%20AOS%20Voice%20Products.pdf

This should assist in setting up the routing.

.

[jayh]

• Please post your sanitized configurations.
• Take a look at the voice grouped-trunk configurations and tweak the one toward the PBX to only include those DIDs that appear on the PBX.  It may be that the default match is hitting the PBX.  You probably want to route calls to PBX DIDs directly from the FXS rather than hairpinning via your SIP provider.
• It's possible that the FXS is trying to authenticate or register to the SIP trunk and failing, then falling back to the PBX.  debug voice verbose and debug sip stack messages will show if this is the case.  Do this off-hours so as not to get a lot of chatter. You may need to tweak the SIP identity or caller-ID on the FXS to make the SIP provider happy.
• You may need to do some DNIS match-substitute on the SIP trunk so that the right number of digits are being sent to the SIP provider such as add local area code, add or strip leading 1, etc. so that the person dialing the phone in the normal way sends what is expected to the SIP provider.  Debugs will help analyze this.

[normrr]

Just to clarify, We are the ISP and we provide voip service through the Netsapiens switch. Intermax is a local ISP

and we have been doing voice about 2 years, but ramping up rapidly. Thanks you both for your replies, I looked at

the info and see that i have an issue in binding the traffic to the correct trunk. I am new to scripting the Adtrans and

know I am asking basic questions. This is the base config for the PRI that I need to add the analogs too.

hostname "XXXXXXXXXX"

enable password XXXXXXXX

!

clock timezone -8

!

ip subnet-zero

ip classless

ip routing

!

!

domain-proxy

name-server xxx.xx.xx.xxx x.x.x.x

!

!

no auto-config

!

event-history on

no logging forwarding

no logging email

!

no service password-encryption

!

username "XXXXXX" password "XXXXXXX"

!

banner motd #

        Important 

Web username/password is configured to admin/password.

Enable and Telnet passwords are configured to "password".

Please change them immediately.

The ethernet 0/1 interface is enabled with an address of 10.10.10.1

Telnet/SSH access is also enabled.

#

!

!

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

no dot11ap access-point-control

!

!

!

!

!

!

!

!

!

!

!

!

!

!

interface loop 1

  no ip address

  no shutdown

!

interface eth 0/1

  ip address  xxx.xxx.xxx.xxx  255.255.255.xxx

  media-gateway ip primary

  no awcp

  no shutdown

!

!

!

!

interface t1 0/1

  shutdown

!

interface t1 0/2

  description Intermax

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

!

interface pri 1

  description pri 1

  isdn name-delivery setup

  connect t1 0/2 tdm-group 1

  no shutdown

!

!

interface fxs 0/1

  shutdown

!

interface fxs 0/2

  shutdown

!

interface fxs 0/3

  shutdown

!

interface fxs 0/4

  shutdown

!

!

isdn-group 1

  min-channels 1

  max-channels 23

  connect pri 1

!

isdn-number-template 0 prefix "" plan 0 type 4 NXX-XXXX

!

!

!

timing-source internal

!

timing-source internal secondary

!

!

!

!

!

!

ip route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.xxx

!

no tftp server

no tftp server overwrite

http server

http secure-server

no snmp agent

no ip ftp server

no ip scp server

no ip sntp server

!

!

!

!

!

!

!

!

ip sip

ip sip udp 5060

no ip sip tcp

!

!

!

voice feature-mode network

voice forward-mode network

!

!

!

!

!

!

!

!

voice dial-plan 1 local NXX-XXXX

voice dial-plan 2 long-distance MXXX

!

!

!

!

voice codec-list "SIP to PRI"

  default

  codec g711ulaw

  codec g711alaw

  codec g722

  codec g729

!

!

!

voice trunk T01 type sip

  description "SIP to Intermax"

  sip-server primary xxxx.xxxxxxxxxx.com

  registrar primary xxx.xxx.xxx.xxx

  no registrar require-expires

  outbound-proxy primary xxxx.xxxxxxxxxx.com

  domain ""

  dial-string source to

  max-number-calls 23

  codec-list "SIP to PRI" both

  authentication username "XXXXXXXXX" password "XXXXXXXXX"

  no diversion-supported

  transfer-mode local

!

voice trunk T02 type isdn

  description "PRI to PBX"

  resource-selection linear ascending

  connect isdn-group 1

  early-cut-through

  rtp delay-mode adaptive

  codec-list "SIP to PRI"

!

!

voice grouped-trunk "INTERMAX SIP TO PRI"

  trunk T01

  accept 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 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept 10-10-XXX-$ cost 0

  reject 976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

!

!

voice grouped-trunk "PRI TO PBX"

  trunk T02

  accept 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 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept $ cost 0

  accept 10-10-XXX-$ cost 0

  reject 976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

ip sip qos dscp 46

!

!

!

ip rtp symmetric-filter

!

!

!

line con 0

  no login

!

line telnet 0 4

  login

  password xxxxxxxxx

  no shutdown

line ssh 0 4

  login local-userlist

  no shutdown

!

!

!

!

!

end

I have reset the Adtran and reloaded a clean config as my changes had started to create a clutter with no results.

[Anonymous]

Hi normrr:

1.  Enable an FXS interface:

interface fxs 0/1

  no shutdown

2.  Setup a voice user and connect it to an FXS interface:

voice user 10001

  connect fxs 0/1

  first-name "FXS"

  last-name "1"

  password encrypted "****************"

  ! Optionally set codec and features

  modem-passthrough

  codec-list G711-only

  ! Assign a telephone number/DID

  alias 2605551212

3.  Modify your trunk group accept statements.  In your config, the trunk groups overlap digit strings and have the same cost.  Consider the following cost changes so that calls from FXS (at least to non-matched numbers) are forced out to your SIP service.  Last, it's a good idea to explicitly reject numbers that go to voice user (FXS) from the PRI trunk group.

voice grouped-trunk "INTERMAX SIP TO PRI"

  trunk T01

  accept 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 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept 10-10-XXX-$ cost 0

  reject 976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

!

!

voice grouped-trunk "PRI TO PBX"

  trunk T02

  accept $ cost 10

  reject 976-XXXX

  reject 2605551212

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

You're normally safe to omit the 411, 611, etc. entries from the PBX-facing trunk group, as the provider isn't going to send calls to those numbers toward the PBX.  Often, accept $ cost 10 is sufficient on that trunk group.  Accept statements determine what can be sent from the AOS gateway out a given trunk group.  I left your reject statements in place as a block, just in case the PBX has another PSTN service and a 900/976 number is dialed from the FXS.

Let us know if you have questions (and how it goes)!

Chris

[normrr]

First I want to say Thanks to all you guys who have taken time to respond, secondly this is driving me nuts.

I have incorporated all the suggestions that fit my issue, still I can get inbound calls to a phone on the FXS1 port

but when I dial out I activate the D ch signaling on the Meter attached to the PRI. Fast busy when the meter is off.

It seem that I don't have a path from the FXS1 to the SIP for the out bound traffic.

!

clock timezone -8

!

ip subnet-zero

ip classless

ip routing

!

!

ip name-server xxx.xxx.xxx.xxx x.x.x.x

!

!

no auto-config

!

event-history on

no logging forwarding

no logging email

!

no service password-encryption

!

username "xxxx" password "xxxxx"

!

banner motd #

        Important 

Web username/password is configured to admin/password.

Enable and Telnet passwords are configured to "password".

Please change them immediately.

The ethernet 0/1 interface is enabled with an address of 10.10.10.1

Telnet/SSH access is also enabled.

#

!

!

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

no dot11ap access-point-control

!

!

!

!

!

!

!

!

!

!

!

!

!

interface loop 1

  no ip address

  no shutdown

!

interface eth 0/1

  ip address  xxx.xxx.xxx.xxx.  255.255.255.xxx

  media-gateway ip primary

  no awcp

  no shutdown

!

!

!

!

interface t1 0/1

  shutdown

!

interface t1 0/2

  description Intermax

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

!

interface pri 1

  description pri 1

  isdn name-delivery setup

  connect t1 0/2 tdm-group 1

  role network b-channel-restarts disable

  no shutdown

!

!

interface fxs 0/1

  no shutdown

!

interface fxs 0/2

  shutdown

!

interface fxs 0/3

  shutdown

!

interface fxs 0/4

  shutdown

!

!

isdn-group 1

  min-channels 1

  max-channels 23

  connect pri 1

!

isdn-number-template 0 prefix "" plan 0 type 4 NXX-XXXX

!

!

!

timing-source internal

!

timing-source internal secondary

!

!

!

!

!

!

!

ip route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.xxx

!

no ip tftp server

no ip tftp server overwrite

ip http server

no ip http secure-server

no ip snmp agent

no ip ftp server

no ip scp server

no ip sntp server

!

!

!

!

!

!

ip sip

ip sip udp 5060

no ip sip tcp

!

!

!

voice feature-mode network

voice forward-mode network

!

!

!

!

!

!

!

voice dial-plan 1 local NXX-XXXX

voice dial-plan 2 long-distance MXXX

!

!

!

!

voice codec-list "SIP to PRI"

  default

  codec g711ulaw

  codec g711alaw

  codec g722

  codec g729

!

!

!

voice trunk T01 type sip

  description "SIP to Intermax"

  sip-server primary voip.xxxxxxxxxxxx.com

  registrar primary xxx.xxx.xxx.xxx

  no registrar require-expires

  outbound-proxy primary voip.xxxxxxxxxx.com

  authentication username "xxxxxxxx" password "xxxxxxxx"

  dial-string source to

  max-number-calls 23

  no diversion-supported

  transfer-mode local

!

voice trunk T02 type isdn

  description "PRI to PBX"

  resource-selection linear ascending

  connect isdn-group 1

  rtp delay-mode adaptive

!

!

voice grouped-trunk "INTERMAX SIP TO PRI"

  trunk T01

  accept 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 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept 10-10-XXX-$ cost 0

  reject 976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

!

!

voice grouped-trunk "PRI TO PBX"

  trunk T02

  accept $ cost 10

  reject 976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

  reject xxxxxxxxx  (same as the alias)

!

!

voice grouped-trunk T01

!

!

voice user 400

  connect fxs 0/1

  first-name "FSX"

  last-name "1"

  password "1234"

  codec-group "SIP to PRI"

  alias xxxxxxxxx

!

!

!

!

!

!

!

!

!

!

!

!

!

!

ip sip proxy domain "voip.xxxxxxxxxxx.com"

ip sip proxy grammar request-uri host domain

ip sip proxy grammar to host domain

ip sip proxy grammar from host domain

!

!

!

!

!

!

!

ip sip qos dscp 46

!

!

ip rtp symmetric-filter

!

!

!

line con 0

  no login

!

line telnet 0 4

  login

  password !NMnet09

  no shutdown

line ssh 0 4

  login local-userlist

  no shutdown

!

!

!

!

!

end

Thanks again all!

[Anonymous]

normrr,

    One recommendation I would add to that is changing your voice grouped-trunk "PRI TO PBX" to only accept numbers that actually belong on the PRI instead of doing accept $ (ALL) cost 10.  I would list every number with a cost of 0 no generic entries.  This list is used by the switchboard of the system to help determine call routing so whena call comes in on the SIP trunk the switchboard compares the dialed number to all other trunks on the system looking for the best match with the lowest cost and then assigns a score to each possible route it finds and the route with the best score gets the call.  An exact match always gets a better value then a generic match.  This gets very important when you start getting into systems with multiple PRIs that have preferred routes for example you could have two PRIs where PRI 1 would be mainly for incoming calls and PRI 2 is designated with outgoing calls yet they have mutual failover incase one or the other is down or full.  In this scenario you would assign each did on both list but give PRI 1 a cost of 0 and the second PRI a cost 100 so inbound calls will always go to PRI 1 first.  When you are making outbound calls from the PRI the switchboard would look at the accept lists for the trunks other then the PRI list so you would want the generic accepts there so the dialed number will match that trunk.  Either method will work but I think the extra security of ensuring only valid calls are sent to the PBX is worth the extra effort.

John Wable

[normrr]

Thanks John, and everyone who helped out. I will only use the analog FXS for FAX, would adding a reject to the "PRI TO PBX" for the FAX numbers work as well as listing the numbers on the PRI. Just asking as we have PRI's with 200 - 300 DID's. And I have the Analog ports working now with good in and outbound audio.

Thanks again all. I will be back with other issues I am sure.

[Anonymous]

Using 'reject' on the PRI TO PBX group for FXS user DIDs is a great way to go.

[normrr]

Thanks CJ!. It just seems a little easier, as long as the results are the same.