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incoming call problems with 2 PSTN sip trunks

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I have a scenario where I am attempting redundant FXS registrations to two separate trunks (T01 and T02).

The registrations are working with success.

I would like to allow access for ingress calls from both of the trunks (T01 and T02) to complete to the FXS voice user lines.

My egress Termination call scenarios seem to route as desired, first to T01 then to T02 if non-response, 503 etc…with success.

However, if ingress calls come in from T01 they work, but if calls are ingress from T02, I get the following error:

15:57:02.648 VOICE.SUMMARY T02 is calling T01 (xxxxxxxxx).

15:57:02.649 VOICE.SUMMARY Call from T02 to T01 (xxxxxxxxxx) ended by T01: resource unavailable

  1. 2019.08.08 15:57:02 SB.CALL 130 RTP resource unavailable or SDP negotiation failed. Call from (xxxxxxxxxxxx) to (xxxxxxxxxxx).

I am trying to build a config that will allow the voice users to be registered to both T01 and T02 simultaneously.

In addition to this set up, I need both the ingress and egress calls to utilize both trunks in a redundant failover manner.

Please see current config below and advise on a best practices solution:

TEST-ADT-01#sh run

Building configuration...

!

!

! ADTRAN, Inc. OS version R13.5.1.E

! Boot ROM version R10.9.3.B1

! Platform: Total Access 908e (3rd Gen), part number

! Serial number

!

!

hostname "TEST-ADT-01"

enable password encrypted xxxxx

!

!

clock timezone -5-Eastern-Time

!

ip subnet-zero

ip classless

ip default-gateway 10.255.220.1

ip routing

ipv6 unicast-routing

!

!

name-server X.X.X.X x.x.x.x

!

!

no auto-config

auto-config authname adtran encrypted password xxxxxx

!

event-history on

no logging forwarding

no logging email

!

service password-encryption

!

username "XXXXXXXXX" password encrypted "xxxxxxxxxxxxxxxxx"

!

!

ip firewall

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

!

!

!

no dot11ap access-point-control

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

interface eth 0/1

  encapsulation 802.1q

  no shutdown

!

!

interface eth 0/1.220

  vlan-id 220

  ip address 10.255.220.20  255.255.255.0

  media-gateway ip primary

  no shutdown

!

interface eth 0/2

  ip address 192.168.66.95  255.255.255.0

  no shutdown

  media-gateway ip primary

!

!

!

interface gigabit-eth 0/1

  no ip address

  shutdown

!

!

!

!

interface t1 0/1

  shutdown

!

interface t1 0/2

  shutdown

!

interface t1 0/3

  shutdown

!

interface t1 0/4

  shutdown

!

!

interface fxs 0/1

  no shutdown

!

interface fxs 0/2

  no shutdown

!

interface fxs 0/3

  no shutdown

!

interface fxs 0/4

  description "OSN Test DID - +xxxxxxxxxxx"

  no shutdown

!

interface fxs 0/5

  description "OSN Test DID - +xxxxxxxxxx"

  no shutdown

!

interface fxs 0/6

  no shutdown

!

interface fxs 0/7

  no shutdown

!

interface fxs 0/8

  no shutdown

!

!

!

!

!

!

!

ip access-list standard MGMT

  permit X.X.X.X 0.0.0.255

  permit X.X.X.X 0.0.0.7

  permit X.X.X.X 0.0.0.31

  permit X.X.X.X 0.0.0.255

  permit X.X.X.X 0.0.0.255

  permit X.X.X.X 0.0.0.255

  permit X.X.X.X 0.0.0.255

!

!

!

!

!

!

ip route 0.0.0.0 0.0.0.0 10.255.220.1

!

no tftp server

no tftp server overwrite

http server

http secure-server

snmp agent

no ip ftp server

no ip scp server

no ip sntp server

!

!

!

!

snmp-server community xxxx RO

!

!

!

!

sip

sip udp 5060

no sip tcp

no sip tls

!

!

voice international-prefix abbreviated

!

voice feature-mode network

voice timeouts interdigit 5

voice flashhook mode transparent

voice forward-mode network

voice num-rings 9

!

!

!

!

voice spre 1 *XX

!

!

!

!

voice dial-plan 2 local N11

voice dial-plan 3 local 1-NXX-NXX-XXXX

voice dial-plan 4 international 011XXXXXXXXXXXX

voice dial-plan 5 international 00XXXXXXXXXXXX

!

!

!

!

voice class-of-service GLOBAL

  call-privilege all

!

voice codec-list GLOBAL

  default

  codec g711ulaw

  codec g729

!

!

!

voice trunk T01 type sip

  description "AAAAA"

  sip-server primary X.X.X.X

  registrar primary X.X.X.X

  conferencing-uri "t"

  max-number-calls 10

  codec-list GLOBAL both

  update-supported

!

voice trunk T02 type sip

  description "BBBBB"

  sip-server primary Y.Y.Y.Y

  registrar primary Y.Y.Y.Y

  conferencing-uri "t"

  max-number-calls 10

  codec-list GLOBAL both

  update-supported

!

!

voice grouped-trunk AAAAA

  description "Connection to AAAAA (Ingress/Egress)"

  trunk T01

  accept $ cost 0

  accept XXXXXXXXXXX cost 0

  accept XXXXXXXXXX cost 0

!

!

voice grouped-trunk BBBBB

  description "Connection to BBBBB (Ingress/Egress)"

  trunk T02

  accept $ cost 0

  accept XXXXXXXXXXX cost 0

  accept XXXXXXXXXX cost 0

!

!

voice userYYYYYYYYYYY

  connect fxs 0/4

  cos "GLOBAL"

  password encrypted "xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx"

  no call-waiting

  did "xxxxxxxxxxx"

  no special-ring-cadences

  no message-waiting

  sip-identity xxxxxxxxxxx T01 register auth-name "xxxxxxxxxxx" password encrypted "aaaaaaaaaaaaaaaaaaaaaaa"

  sip-identity xxxxxxxxxxx T02 register auth-name "xxxxxxxxxxx" password encrypted "bbbbbbbbbbbbbbbbbbbbbbbb"

  sip-authentication password encrypted "bbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbb"

  modem-passthrough

  t38

  no echo-cancellation

  rtp dtmf-relay inband

  codec-list GLOBAL

!

!

!

voice user zzzzzzzzzzzzzzzz

  connect fxs 0/5

  cos "GLOBAL"

  password encrypted "zzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzz"

  description "+zzzzzzzzzzzzzz"

  no call-waiting

  did "zzzzzzzzzz"

  no special-ring-cadences

  no message-waiting

  sip-identity 6498010224 T01 register auth-name "zzzzzzzzzz" password encrypted "zzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzz"

  sip-identity 6498010224 T02 register auth-name "zzzzzzzzzz" password encrypted "xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx"

  sip-authentication password encrypted "yyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy"

  modem-passthrough

  t38

  no echo-cancellation

  rtp dtmf-relay inband

  codec-list GLOBAL

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

sip session-timer

sip session-timer min-se 1800

!

!

!

!

!

!

!

!

line con 0

  login

!

line telnet 0 4

  login

  password encrypted xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx

  no shutdown

  ip access-class MGMT in

line ssh 0 4

  login local-userlist

  no shutdown

!

!

ntp peer pool.ntp.org

!

!

!

end

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Honored Contributor
Honored Contributor

Re: incoming call problems with 2 PSTN sip trunks

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What I would do is to set up a registered SIP trunk on the Adtran itself to both providers.

voice trunk T01 type sip

  description "AAAAA"

  sip-server primary X.X.X.X

  registrar primary X.X.X.X

  register YYYYYYYYYY auth-name "zzzzzzzzz" password "zzzzzzzz"

  register ZZZZZZZZZZ auth-name "zzzzzzzzz" password "zzzzzzzz"

  conferencing-uri "t"

  max-number-calls 10

  codec-list GLOBAL both

  update-supported

voice trunk T02 type sip

  description "BBBBB"

  sip-server primary X.X.X.X

  registrar primary X.X.X.X 

  register YYYYYYYYYY auth-name "zzzzzzzzz" password "zzzzzzzz"

  register ZZZZZZZZZZ auth-name "zzzzzzzzz" password "zzzzzzzz"

  conferencing-uri "t"

  max-number-calls 10

  codec-list GLOBAL both

  update-supported

The YYYYYYYYYY and ZZZZZZZZZZ in the register commands above are the DIDs of your FXS users, with appropriate auth-name and password.

Then delete the sip-identity and registration from the users themselves.

For scalability if you have more than one or two FXS users, inquire with your carriers about having a registered trunk instead of registering individual DIDs. This way you can add users without having to register all of them individually. The trunk uses a single "pilot" number to register and the SIP provider routes all of your DIDs down the same trunk.

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Honored Contributor
Honored Contributor

Re: incoming call problems with 2 PSTN sip trunks

Jump to solution

What I would do is to set up a registered SIP trunk on the Adtran itself to both providers.

voice trunk T01 type sip

  description "AAAAA"

  sip-server primary X.X.X.X

  registrar primary X.X.X.X

  register YYYYYYYYYY auth-name "zzzzzzzzz" password "zzzzzzzz"

  register ZZZZZZZZZZ auth-name "zzzzzzzzz" password "zzzzzzzz"

  conferencing-uri "t"

  max-number-calls 10

  codec-list GLOBAL both

  update-supported

voice trunk T02 type sip

  description "BBBBB"

  sip-server primary X.X.X.X

  registrar primary X.X.X.X 

  register YYYYYYYYYY auth-name "zzzzzzzzz" password "zzzzzzzz"

  register ZZZZZZZZZZ auth-name "zzzzzzzzz" password "zzzzzzzz"

  conferencing-uri "t"

  max-number-calls 10

  codec-list GLOBAL both

  update-supported

The YYYYYYYYYY and ZZZZZZZZZZ in the register commands above are the DIDs of your FXS users, with appropriate auth-name and password.

Then delete the sip-identity and registration from the users themselves.

For scalability if you have more than one or two FXS users, inquire with your carriers about having a registered trunk instead of registering individual DIDs. This way you can add users without having to register all of them individually. The trunk uses a single "pilot" number to register and the SIP provider routes all of your DIDs down the same trunk.

View solution in original post

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