1 Reply Latest reply on Aug 8, 2019 4:00 PM by jayh

    incoming call problems with 2 PSTN sip trunks

    lgordnier_osn New Member

      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

        • Re: incoming call problems with 2 PSTN sip trunks
          jayh Hall_of_Fame

          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.