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New Contributor III

TA924 to Asterisk SIP?

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This should seem straightforward, but I cannot get outcalling from my TA924 FXS ports through my Asterisk to work. Inbound calling through the Asterisk to these FXS ports works just fine. But when I grab a line from one of my FXS ports, the Asterisk just comes back with the automated message that "Your call cannot be completed as dialed." My TA924 is set with T01 being the only trunk, connecting to my Asterisk for SIP via the TA924's Ethernet port. The TA924 doesn't have PRI connectivity, it only has the FXS ports and then the Ethernet port for communicating with the LAN. The Asterisk itself has the SIP trunks defined for PSTN access. And SIP clients other than the ones on the TA924 are able to take/make PSTN calls just fine.

Here is the config defined as my TA924. It's all LAN-based private IP's between the TA924 and the Asterisk, so I can't see where NAT would come into play. I have detailed debug logs from both endpoints, which I can share as well. Just wondering what could be going on looking at the TA924's config.

Any suggestions?

!

!

! ADTRAN, Inc. OS version 14.04.00.E

! Boot ROM version 12.06.00

! Platform: Total Access 924, part number 4210924L1

! Serial number LBADTN0634AF158

!

!

hostname "TA924"

enable password password

!

clock timezone -5-Eastern-Time

!

ip subnet-zero

ip classless

ip domain-proxy

ip domain-name "diamondcellar.local"

ip routing

!

no auto-config

!

event-history on

no logging forwarding

no logging email

logging email priority-level info

!

no service password-encryption

!

username "admin" password "password"

!

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 h323

!

!

!

!

!

!

!

!

!

!

!

!

interface eth 0/1

  ip address dhcp

  media-gateway ip primary

  no shutdown

!

!

!

!

interface t1 0/1

  no shutdown

!

interface t1 0/2

  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

!

interface fxs 0/17

  no shutdown

!

interface fxs 0/18

  no shutdown

!

interface fxs 0/19

  no shutdown

!

interface fxs 0/20

  no shutdown

!

interface fxs 0/21

  no shutdown

!

interface fxs 0/22

  no shutdown

!

interface fxs 0/23

  no shutdown

!

interface fxs 0/24

  no shutdown

!

!

!

!

!

!

!

!

no ip tftp server

no ip tftp server overwrite

ip http server

ip http secure-server

no 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 codec-list "Default Codecs"

  default

  codec g711ulaw

!

!

voice trunk T01 type sip

  description "iPBX Asterisk Server"

  sip-server primary 10.0.0.164 udp 5160

  registrar primary 10.0.0.164 udp 5160

  outbound-proxy primary 10.0.0.164 udp 5160

  conferencing-uri "10.0.0.164"

!

!

voice grouped-trunk DEFAULT

  no description

  trunk T01

  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 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept 0-NXX-NXX-XXXX cost 0

  accept 10-10-XXX-$ cost 0

  reject NXX-976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

!

!

!

!

voice user 607

  connect fxs 0/8

  first-name "Easton Sales"

  last-name "Cordless"

  password "1234"

  no special-ring-cadences

  sip-identity 607 T01 register auth-name "607" password "D1am0nd5"

  codec-group "Default Codecs"

!

!

voice user 608

  connect fxs 0/7

  first-name "EDC"

  last-name "5"

  password "1234"

  no special-ring-cadences

  sip-identity 608 T01 register auth-name "608" password "D1am0nd5"

  modem-passthrough

  no echo-cancellation

!

!

voice user 609

  connect fxs 0/9

  first-name "Easton Shop"

  last-name "Cordless"

  password "1234"

  no special-ring-cadences

  sip-identity 609 T01 register auth-name "609" password "D1am0nd5"

!

!

voice user 641

  connect fxs 0/2

  first-name "Modem"

  last-name "1"

  password "1234"

  no special-ring-cadences

  sip-identity 641 T01 register auth-name "641" password "D1am0nd5"

  modem-passthrough

  no echo-cancellation

!

!

voice user 644

  connect fxs 0/10

  first-name "Easton Main"

  last-name "Fax"

  password "1234"

  no special-ring-cadences

  sip-identity 644 T01 register auth-name "644" password "D1am0nd5"

  modem-passthrough

  no echo-cancellation

!

!

voice user 645

  connect fxs 0/3

  first-name "Easton Fax"

  last-name "2"

  password "1234"

  no special-ring-cadences

  sip-identity 645 T01 register auth-name "645" password "D1am0nd5"

  modem-passthrough

  no echo-cancellation

!

!

voice user 651

  connect fxs 0/11

  first-name "Easton"

  last-name "Vault"

  password "1234"

  no special-ring-cadences

  sip-identity 651 T01 register auth-name "651" password "D1am0nd5"

!

!

voice user 652

  connect fxs 0/1

  first-name "EDC"

  last-name "1"

  password "1234"

  no special-ring-cadences

  sip-identity 652 T01 register auth-name "652" password "D1am0nd5"

  modem-passthrough

  no echo-cancellation

!

!

voice user 653

  connect fxs 0/4

  first-name "EDC"

  last-name "2"

  password "1234"

  no special-ring-cadences

  sip-identity 653 T01 register auth-name "653" password "D1am0nd5"

  modem-passthrough

  no echo-cancellation

!

!

voice user 654

  connect fxs 0/5

  first-name "EDC"

  last-name "3"

  password "1234"

  no special-ring-cadences

  sip-identity 654 T01 register auth-name "654" password "D1am0nd5"

  modem-passthrough

  no echo-cancellation

!

!

voice user 655

  connect fxs 0/6

  first-name "EDC"

  last-name "4"

  password "1234"

  no special-ring-cadences

  sip-identity 655 T01 register auth-name "655" password "D1am0nd5"

  modem-passthrough

  no echo-cancellation

!

!

!

!

!

ip sip

no ip sip registrar authenticate

!

ip sip grammar request-uri host domain

ip sip grammar from host local

ip sip grammar p-asserted-identity host domain

ip sip grammar to host domain

!

!

!

!

line con 0

  no login

!

line telnet 0 4

  login

  password password

  no shutdown

line ssh 0 4

  login local-userlist

  no shutdown

!

sntp server pool.ntp.org

!

end

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New Contributor III

Re: TA924 to Asterisk SIP?

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This can be closed. I forgot to include proper outbound routes on my Asterisk. Since it is a centralized IP-PBX, each remote site has a different extension range. This is conditionally handled for outbound routing out the same SIP trunks. I just have to conditionally handle outbound CID, MoH, etc. Once I added this new route I am working fine. The TA924 is working fine!

View solution in original post

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1 Reply
Highlighted
New Contributor III

Re: TA924 to Asterisk SIP?

Jump to solution

This can be closed. I forgot to include proper outbound routes on my Asterisk. Since it is a centralized IP-PBX, each remote site has a different extension range. This is conditionally handled for outbound routing out the same SIP trunks. I just have to conditionally handle outbound CID, MoH, etc. Once I added this new route I am working fine. The TA924 is working fine!

View solution in original post

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Reply