15 Replies Latest reply on Apr 11, 2019 10:16 PM by g-man

    Inbound calls not making it to softswitch

    g-man New Member

      I am hoping this will be my final question on this specific journey. I have a TA900 that is currently serving an older PBX with PRI. I have also been working on installing a soft switch behind the TA900. I am able to place outbound calls from the softswitch but inbound calls are not making it. Initially I was seeing a 404 error on the adtran but after playing with the config some more I lost that as well? Luckily PRI continues to work as expected!

       

      I tried adding another the following for inbound calls with no success. Not sure how to register a trunk from the PBX to Adtran either.

       

      Voice Trunk T03 type SIP

      sip-server primary 192.168.33.1

      transfer-mode network

      grammer from host local

       

      Voice grouped-Trunk PBX

      accept 13235556666 cost 0

       

       

       

      Current Config

      !

      !

      clock timezone -8

      !

      ip subnet-zero

      ip classless

      ip routing

      ipv6 unicast-routing

       

       

      auto-config

      !

       

       

      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

      description WAN

      ip address  76.10.76.10  255.255.255.248

      ip address  76.10.76.11  255.255.255.255  secondary

      ip access-policy Public

      media-gateway ip primary

      no shutdown

      !

      !

      interface eth 0/2

      description  (LAN)

      ip address  192.168.33.1  255.255.255.0

      ip access-policy Private

      media-gateway ip primary

      no awcp

      no shutdown

      !

      !

      !

      interface gigabit-eth 0/1

        no ip address

        shutdown

      !

      !

      !

      !

      interface t1 0/1

        shutdown

      !

      interface t1 0/2

        shutdown

      !

      interface t1 0/3

        lbo short 15

        tdm-group 1 timeslots 1-24 speed 64

        no shutdown

      !

      interface t1 0/4

        shutdown

      !

      !

      interface pri 1

        isdn name-delivery proceeding

        connect t1 0/3 tdm-group 1

        digits-transferred 4

        no shutdown

      !

      !

      interface fxs 0/1

        impedance 600r

        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

      !

      !

      isdn-group 1

        connect pri 1

      !

      !

      !

      !

      !

      !

      !

      !

      ip access-list standard allow-all

        remark allow all traffic

        permit any

      !

      ip access-list standard mgmt-allow-list

        70.11.11.99

      !

      ip access-list standard sip-allow-list

        permit hostname xx.com

       

       

      !

      !

      ip access-list extended WEB-ACL-3

        permit tcp any  any eq https 

        permit tcp any  any eq ssh 

      !

      ip access-list extended WEB-ACL-4

      remark 1:1 NAT 76.10.76.11 > 192.168.33.11

      permit ip any  host 76.10.76.11 

      !

      ip access-list extended WEB-ACL-5

        remark 1:1 NAT 192.168.33.11 > 76.10.76.11

        permit ip host 192.168.33.11  any   

      !

      !

      ip policy-class Private

        nat source list allow-all interface eth 0/1 overload policy Public

        allow list allow-all self

        nat source list WEB-ACL-5 address 76.10.76.11 overload

      !

      ip policy-class Public

        nat destination list WEB-ACL-4 address 192.168.33.11

        allow list allow-all self

        allow list WEB-ACL-3 self

      !

      !

      !

      ip route 0.0.0.0 0.0.0.0 76.10.76.10

      !

      no tftp server

      no tftp server overwrite

      no http server

      http secure-server

      no snmp agent

      no ip ftp server

      no ip scp server

      no ip sntp server

      !

      !

       

       

      sip

      sip udp 5060

      no sip tcp

      !

      !

      !

      voice feature-mode network

      voice forward-mode network

      !

      !

       

       

      voice dial-plan 2 long-distance 1-NXX-NXX-XXXX

      !

      !

      !

      !

      voice codec-list VOICE

        default

        codec g711ulaw

      !

      voice codec-list FAX

        codec g711ulaw

      !

      !

      !

      voice trunk T01 type sip

        description "SIP"

        match dnis "91-NXX-NXX-XXXX" substitute "1-NXX-NXX-XXXX"

        match dnis "9NXX-XXXX" substitute "1-310-NXX-XXXX"

        match dnis "NXX-NXX-XXXX" substitute "1-NXX-NXX-XXXX"

        match dnis "NXX-XXXX" substitute "1-310-NXX-XXXX"

        sip-server primary 88.88.88.88

        registrar primary 88.88.88.88

        domain "76.10.76.10"

        register 100XXXXXXX auth-name "" password ""

        codec-list VOICE both

        authentication username "" password ""

      !

      voice trunk T02 type isdn

        description "DSX-1"

        resource-selection linear ascending

        connect isdn-group 1

        no early-cut-through

        match dnis "1800XXXXXXX" substitute "13235551212"

        match dnis "1844XXXXXXX" substitute "13235551212"

        rtp delay-mode adaptive

        codec-list VOICE

      !

      !

      voice grouped-trunk SIP

        trunk T01

        accept $ cost 0

      !

      !

      !

      !

      voice grouped-trunk ISDN

        trunk T02

        accept 1323XXXXXXX cost 0

      !

      sip access-class ip "sip-allow-list" in

      !

      line con 0

        no login

      !

      line telnet 0 4

        login local-userlist

        password password

        shutdown

        ip access-class mgmt-allow-list in

      line ssh 0 4

        login local-userlist

        no shutdown

        ip access-class mgmt-allow-list in

      !

      !

      !

      !

      !

      end

        • Re: Inbound calls not making it to softswitch
          g-man New Member

          I was able to get the log of my calls and it looks like 76.10.76.11 which is the secondary address is not responding to invites. Shouldn't

           

          ip access-list standard sip-allow-list

            permit hostname xx.com

           

          allow the sip traffic to the switch?

            • Re: Inbound calls not making it to softswitch
              jayh Hall_of_Fame

              g-man wrote:

               

              I was able to get the log of my calls and it looks like 76.10.76.11 which is the secondary address is not responding to invites. Shouldn't

               

              ip access-list standard sip-allow-list

              permit hostname xx.com

               

              allow the sip traffic to the switch?

              OK, it looks like there are a couple of oddities here.

               

              Your softswitch is on the LAN with an address of 192.168.33.XX, correct?

              Add that private IP to your sip-allow-list .

               

              You have trunk T03 pointing to your own interface as the SIP server. Point it to the IP of your softswitch. On the softswitch, point its SIP server address to 192.168.33.1.

               

              Also note that a few firmware revisions back, Adtran deliberately broke the ability of TA900 devices to process some SIP-to-SIP calls unless you purchase an extra SBC license so you may need that to move forward. It's relatively cheap but quite an annoyance.

                • Re: Inbound calls not making it to softswitch
                  g-man New Member

                  I inquired with adtran about the license. It seems like something is blocking my invite from the provider yet I have other calls that are processing?

                   

                  Routing a call from one SIP trunk to another SIP trunk does require an SBC license, however if the ADTRAN receives an INVITE, it will at least acknowledge the packet, as long as the request-URI has the ADTRAN's IP address (addressed to the ADTRAN). If the SIP packet is not addressed to the ADTRAN, then the ADTRAN will not respond.

              • Re: Inbound calls not making it to softswitch
                jayh Hall_of_Fame

                g-man wrote:

                 

                 

                 

                I tried adding another the following for inbound calls with no success. Not sure how to register a trunk from the PBX to Adtran either.

                 

                Voice Trunk T03 type SIP

                sip-server primary 192.168.33.1

                transfer-mode network

                grammer from host local

                 

                Voice grouped-Trunk PBX

                accept 13235556666 cost 0

                 

                Try adding:

                 

                Voice grouped-Trunk PBX

                trunk T03

                accept 13235556666 cost 0

                  • Re: Inbound calls not making it to softswitch
                    g-man New Member

                    jayh,

                     

                    Thank you so much for the assistance. I did as you suggested and still nothing. What is odd is that the call is not even making it to the TA. I tried SIP debug but I do not see anything. When I ask the SIP provider they say they are not getting a response to the invite. I would at least expect to see something. I am sending calls to the secondary address on the TA. Would the license be causeing such an isue

                      • Re: Inbound calls not making it to softswitch
                        jayh Hall_of_Fame

                        g-man wrote:

                         

                        jayh,

                         

                        Thank you so much for the assistance. I did as you suggested and still nothing. What is odd is that the call is not even making it to the TA. I tried SIP debug but I do not see anything. When I ask the SIP provider they say they are not getting a response to the invite.

                        Is this the same SIP provider that is sending calls to the PRI, and is the provider sending them in the same manner? Don't use the secondary address, use the main interface address just as you to for calls to the PRI.

                         

                        What happens if you configure the grouped-trunk on the softswitch T03 to accept one of the numbers now routed to the PRI? Does that call now go to the softswitch?

                          • Re: Inbound calls not making it to softswitch
                            g-man New Member

                            I just tried that exact thing and the call does not make it to the softswitch. I don't even see the call hit the adtran. I pointed the softswitch calls to the Public IP of the secondary interface, the PBX is going to the first.

                            • Re: Inbound calls not making it to softswitch
                              g-man New Member

                              I just got an invite from the PRI side.

                               

                              09:40:42.181 SIP.STACK MSG     Rx: UDP src=88.88.88.88:5060 dst=76.10.76.11:5060
                              09:40:42.181 SIP.STACK MSG         INVITE sip:13237360311@76.10.76.11 SIP/2.0
                              09:40:42.182 SIP.STACK MSG         Via: SIP/2.0/UDP 88.88.88.88:5060;branch=z9hG4bKcvpnng008g81vf0fn7k0.1
                              09:40:42.182 SIP.STACK MSG         From: "Bernard,St." <sip:+16264919734@88.88.88.88>;tag=SDphks201-gK0458d68d
                              09:40:42.182 SIP.STACK MSG         To: <sip:13237360311@76.10.76.11>
                              09:40:42.182 SIP.STACK MSG         Remote-Party-ID: "Bernard,St." <sip:+16264919734@88.88.88.88:5060>;privacy=off;screen=yes
                              09:40:42.182 SIP.STACK MSG         Call-ID: SDphks201-7282d27c2edb07ed73bf4900e7e32ad7-523g533
                              09:40:42.182 SIP.STACK MSG         CSeq: 984876 INVITE
                              09:40:42.182 SIP.STACK MSG         Max-Forwards: 68
                              09:40:42.183 SIP.STACK MSG         Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
                              09:40:42.183 SIP.STACK MSG         Accept: application/sdp
                              09:40:42.183 SIP.STACK MSG         Contact: <sip:88.88.88.88:5060;did=72b.0dc32b76;transport=udp>
                              09:40:42.183 SIP.STACK MSG         Supported: replaces
                              09:40:42.183 SIP.STACK MSG         Content-Length: 254
                              09:40:42.183 SIP.STACK MSG         Content-Disposition: session; handling=required
                              09:40:42.184 SIP.STACK MSG         Content-Type: application/sdp
                              09:40:42.184 SIP.STACK MSG
                              09:40:42.184 SIP.STACK MSG         v=0
                              09:40:42.184 SIP.STACK MSG         o=- 3788767953 759660 IN IP4 99.99.99.99
                              09:40:42.184 SIP.STACK MSG         s=-
                              09:40:42.184 SIP.STACK MSG         c=IN IP4 99.99.99.99
                              09:40:42.184 SIP.STACK MSG         t=0 0
                              09:40:42.185 SIP.STACK MSG         m=audio 4666 RTP/AVP 0 18 101
                              09:40:42.185 SIP.STACK MSG         a=rtpmap:0 PCMU/8000
                              09:40:42.185 SIP.STACK MSG         a=rtpmap:18 G729/8000
                              09:40:42.185 SIP.STACK MSG         a=fmtp:18 annexb=no
                              09:40:42.185 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000
                              09:40:42.185 SIP.STACK MSG         a=fmtp:101 0-15
                              09:40:42.185 SIP.STACK MSG         a=sendrecv
                              09:40:42.186 SIP.STACK MSG         a=ptime:20
                              09:40:42.186 SIP.STACK MSG
                              09:40:42.190 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=88.88.88.88:5060
                              09:40:42.190 SIP.STACK MSG         SIP/2.0 100 Trying
                              09:40:42.190 SIP.STACK MSG         From: "Bernard,St."<sip:+16264919734@88.88.88.88>;tag=SDphks201-gK0458d68d
                              09:40:42.190 SIP.STACK MSG         To: <sip:13237360311@76.10.76.11>
                              09:40:42.190 SIP.STACK MSG         Call-ID: SDphks201-7282d27c2edb07ed73bf4900e7e32ad7-523g533
                              09:40:42.190 SIP.STACK MSG         CSeq: 984876 INVITE
                              09:40:42.191 SIP.STACK MSG         Via: SIP/2.0/UDP 88.88.88.88:5060;branch=z9hG4bKcvpnng008g81vf0fn7k0.1
                              09:40:42.191 SIP.STACK MSG         Contact: <sip:13237360311@76.10.76.11:5060;transport=UDP>
                              09:40:42.191 SIP.STACK MSG         Supported: 100rel,replaces
                              09:40:42.191 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
                              09:40:42.191 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
                              09:40:42.191 SIP.STACK MSG         Content-Length: 0
                              09:40:42.192 SIP.STACK MSG
                              09:40:42.197 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
                              09:40:42.197 SIP.STACK MSG         INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0
                              09:40:42.198 SIP.STACK MSG         From: "Bernard,St." <sip:6264919734@76.10.76.11:5060;transport=UDP>;tag=63139890-a0a5801-13c4-1401a-2ea8031f-1401a
                              09:40:42.198 SIP.STACK MSG         To: <sip:13237360311@192.168.33.3:5060>
                              09:40:42.198 SIP.STACK MSG         Call-ID: 63161588-a0a5801-13c4-1401a-113af5b4-1401a@76.10.76.11
                              09:40:42.198 SIP.STACK MSG         CSeq: 1 INVITE
                              09:40:42.198 SIP.STACK MSG         Via: SIP/2.0/UDP 76.10.76.11:5060;branch=z9hG4bK-1401a-4e266b6-3abb03d
                              09:40:42.198 SIP.STACK MSG         Max-Forwards: 70
                              09:40:42.199 SIP.STACK MSG         Supported: 100rel,replaces
                              09:40:42.199 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
                              09:40:42.199 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
                              09:40:42.199 SIP.STACK MSG         Contact: <sip:6264919734@76.10.76.11:5060;transport=UDP>
                              09:40:42.199 SIP.STACK MSG         Content-Type: application/sdp
                              09:40:42.199 SIP.STACK MSG         Content-Length: 234
                              09:40:42.199 SIP.STACK MSG
                              09:40:42.200 SIP.STACK MSG         v=0
                              09:40:42.200 SIP.STACK MSG         o=- 3788767953 759660 IN IP4 99.99.99.99
                              09:40:42.200 SIP.STACK MSG         s=-
                              09:40:42.200 SIP.STACK MSG         c=IN IP4 99.99.99.99
                              09:40:42.200 SIP.STACK MSG         t=0 0
                              09:40:42.200 SIP.STACK MSG         m=audio 4666 RTP/AVP 0 101
                              09:40:42.200 SIP.STACK MSG         a=sendrecv
                              09:40:42.201 SIP.STACK MSG         a=ptime:20
                              09:40:42.201 SIP.STACK MSG         a=rtpmap:0 PCMU/8000
                              09:40:42.201 SIP.STACK MSG         a=silenceSupp:off - - - -
                              09:40:42.201 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000
                              09:40:42.201 SIP.STACK MSG         a=fmtp:101 0-15
                              09:40:42.201 SIP.STACK MSG
                              09:40:42.702 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
                              09:40:42.702 SIP.STACK MSG         INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0
                              09:40:42.702 SIP.STACK MSG         From: "Bernard,St." <sip:6264919734@76.10.76.11:5060;transport=UDP>;tag=63139890-a0a5801-13c4-1401a-2ea8031f-1401a
                              09:40:42.702 SIP.STACK MSG         To: <sip:13237360311@192.168.33.3:5060>
                              09:40:42.702 SIP.STACK MSG         Call-ID: 63161588-a0a5801-13c4-1401a-113af5b4-1401a@76.10.76.11
                              09:40:42.703 SIP.STACK MSG         CSeq: 1 INVITE
                              09:40:42.703 SIP.STACK MSG         Via: SIP/2.0/UDP 76.10.76.11:5060;branch=z9hG4bK-1401a-4e266b6-3abb03d
                              09:40:42.703 SIP.STACK MSG         Max-Forwards: 70
                              09:40:42.703 SIP.STACK MSG         Supported: 100rel,replaces
                              09:40:42.703 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
                              09:40:42.703 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
                              09:40:42.704 SIP.STACK MSG         Contact: <sip:6264919734@76.10.76.11:5060;transport=UDP>
                              09:40:42.704 SIP.STACK MSG         Content-Type: application/sdp
                              09:40:42.704 SIP.STACK MSG         Content-Length: 234
                              09:40:42.704 SIP.STACK MSG
                              09:40:42.704 SIP.STACK MSG         v=0
                              09:40:42.704 SIP.STACK MSG         o=- 3788767953 759660 IN IP4 99.99.99.99
                              09:40:42.705 SIP.STACK MSG         s=-
                              09:40:42.705 SIP.STACK MSG         c=IN IP4 99.99.99.99
                              09:40:42.705 SIP.STACK MSG         t=0 0
                              09:40:42.705 SIP.STACK MSG         m=audio 4666 RTP/AVP 0 101
                              09:40:42.705 SIP.STACK MSG         a=sendrecv
                              09:40:42.705 SIP.STACK MSG         a=ptime:20
                              09:40:42.705 SIP.STACK MSG         a=rtpmap:0 PCMU/8000
                              09:40:42.706 SIP.STACK MSG         a=silenceSupp:off - - - -
                              09:40:42.706 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000
                              09:40:42.706 SIP.STACK MSG         a=fmtp:101 0-15
                              09:40:42.706 SIP.STACK MSG
                              09:40:43.707 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
                              09:40:43.707 SIP.STACK MSG         INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0
                              09:40:43.707 SIP.STACK MSG         From: "Bernard,St." <sip:6264919734@76.10.76.11:5060;transport=UDP>;tag=63139890-a0a5801-13c4-1401a-2ea8031f-1401a
                              09:40:43.707 SIP.STACK MSG         To: <sip:13237360311@192.168.33.3:5060>
                              09:40:43.707 SIP.STACK MSG         Call-ID: 63161588-a0a5801-13c4-1401a-113af5b4-1401a@76.10.76.11
                              09:40:43.708 SIP.STACK MSG         CSeq: 1 INVITE
                              09:40:43.708 SIP.STACK MSG         Via: SIP/2.0/UDP 76.10.76.11:5060;branch=z9hG4bK-1401a-4e266b6-3abb03d
                              09:40:43.708 SIP.STACK MSG         Max-Forwards: 70
                              09:40:43.708 SIP.STACK MSG         Supported: 100rel,replaces
                              09:40:43.708 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
                              09:40:43.708 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
                              09:40:43.709 SIP.STACK MSG         Contact: <sip:6264919734@76.10.76.11:5060;transport=UDP>
                              09:40:43.709 SIP.STACK MSG         Content-Type: application/sdp
                              09:40:43.709 SIP.STACK MSG         Content-Length: 234
                              09:40:43.709 SIP.STACK MSG
                              09:40:43.709 SIP.STACK MSG         v=0
                              09:40:43.709 SIP.STACK MSG         o=- 3788767953 759660 IN IP4 99.99.99.99
                              09:40:43.709 SIP.STACK MSG         s=-
                              09:40:43.710 SIP.STACK MSG         c=IN IP4 99.99.99.99
                              09:40:43.710 SIP.STACK MSG         t=0 0
                              09:40:43.710 SIP.STACK MSG         m=audio 4666 RTP/AVP 0 101
                              09:40:43.710 SIP.STACK MSG         a=sendrecv
                              09:40:43.710 SIP.STACK MSG         a=ptime:20
                              09:40:43.710 SIP.STACK MSG         a=rtpmap:0 PCMU/8000
                              09:40:43.711 SIP.STACK MSG         a=silenceSupp:off - - - -
                              09:40:43.711 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000
                              09:40:43.711 SIP.STACK MSG         a=fmtp:101 0-15
                              09:40:43.711 SIP.STACK MSG

                              09:40:44.212 SIP.STACK MSG
                              09:40:45.205 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=88.88.88.88:5060
                              09:40:45.205 SIP.STACK MSG         SIP/2.0 503 Service Unavailable
                              09:40:45.205 SIP.STACK MSG         From: "Bernard,St."<sip:+16264919734@88.88.88.88>;tag=SDphks201-gK0458d68d
                              09:40:45.205 SIP.STACK MSG         To: <sip:13237360311@76.10.76.11>;tag=63138c18-a0a5801-13c4-1401d-4ad6b38d-1401d
                              09:40:45.206 SIP.STACK MSG         Call-ID: SDphks201-7282d27c2edb07ed73bf4900e7e32ad7-523g533
                              09:40:45.206 SIP.STACK MSG         CSeq: 984876 INVITE
                              09:40:45.206 SIP.STACK MSG         Via: SIP/2.0/UDP 88.88.88.88:5060;branch=z9hG4bKcvpnng008g81vf0fn7k0.1
                              09:40:45.206 SIP.STACK MSG         Supported: 100rel,replaces
                              09:40:45.206 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
                              09:40:45.206 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
                              09:40:45.206 SIP.STACK MSG         Content-Length: 0

                                • Re: Inbound calls not making it to softswitch
                                  jayh Hall_of_Fame

                                  Your softswitch on 192.168.33.3 isn't responding to the invite, or its response is being filtered.

                                   

                                  From your log the TA900 sent three invites and received no response.

                                   

                                  09:40:42.197 SIP.STACK MSG Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
                                  09:40:42.197 SIP.STACK MSG    

                                  INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0

                                   

                                  09:40:42.702 SIP.STACK MSG Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
                                  09:40:42.702 SIP.STACK MSG     INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0

                                   

                                  09:40:43.707 SIP.STACK MSG Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
                                  09:40:43.707 SIP.STACK MSG     INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0

                                   

                                  After the third attempt the Adtran gave up and sent 503 to your SIP provider.

                                   

                                  09:40:45.205 SIP.STACK MSG Tx: UDP src=76.10.76.11:5060 dst=88.88.88.88:5060
                                  09:40:45.205 SIP.STACK MSG    

                                  SIP/2.0 503 Service Unavailable

                                   

                                  Is either 192.168.33.0 0.0.0.255 or host 192.168.33.3 in your sip-allow-list? It should be.

                                   

                                  Is the softswitch programmed with its SIP server as 192.168.1.1 ? It should be.

                                   

                                  Do you have "media-gateway ip primary" set on your LAN interface of 192.168.33.1? You should.

                                • Re: Inbound calls not making it to softswitch
                                  g-man New Member

                                  Voice Verbose

                                   

                                   

                                  09:54:32.108 TM.T01 01 SipTM_Idle           rcvd SIP call-leg request: INVITE

                                  09:54:32.108 TM.T01 01 SipTM_Idle           call-leg -> Offering

                                  09:54:32.109 TM.T01 01 SipTM_Idle           State change      >> SipTM_Idle->SipTM_Trying

                                  09:54:32.109 TM.T01 01 SipTM_Trying         SDP offer is not loopback request

                                  09:54:32.109 TM.T01 01 SipTM_Trying         Processing From for Caller-ID.

                                  09:54:32.110 TM.T01 01 T01 01 SipTM_Trying         e164 calling number converted to dialstring 13235551212

                                  09:54:32.110 TM.T01 01 SipTM_Trying         Caller ID Name   = "Bernard, St"

                                  09:54:32.110 TM.T01 01 SipTM_Trying         Caller ID Number = "13235551212"

                                  09:54:32.110 TM.T01 01 SipTM_Trying         info: unable to set redirect number(s) from INVITE

                                  09:54:32.110 TM.T01 01 SipTM_Trying         sent: TA->InboundCall

                                  09:54:32.111 TM.T01 01 Looking up source address for destination 69.69.69.69

                                  09:54:32.111 TM.T01 01 call-leg (0x0x63161390) -> src: 76.10.76.10: 5060  dst: 69.69.69.69 : 5060

                                  09:54:32.112 TM.T01 01 SipTM_Trying         sent: 100 Trying

                                  09:54:32.112 TA.T01 01 TAIdle               rcvd: inboundCall from TM

                                  09:54:32.113 TA.T01 01 State change      >> TAIdle->TAInboundCall (TAS_Calling)

                                  09:54:32.113 TA.T01 01 Failed - DID translation: no match for 13237360311, using 13237360311

                                  09:54:32.113 TA.T01 01 TAIdle               sent: call to SB

                                  09:54:32.113 TM.T01 01 SipTM_Trying         tachg -> TAInboundCall

                                  09:54:32.113 TM.T01 01 SipTM_Trying         State change      >> SipTM_Trying->SipTM_Pending

                                  09:54:32.114 SB.CALL 479 Idle                 Called the call routine with 13237360311

                                  09:54:32 SB.TGMgr For dialed number 13237360311, against template $, on TrunkGroup SIP, the score is 500

                                  09:54:32 SB.TGMgr For dialed number 13237360311, against template 13237360311, on TrunkGroup PBX, the score is 12000

                                  09:54:32.114 SB.CCM isMappable:

                                  09:54:32.115 SB.CCM  :  Call Struct 0x0x7520e610 :   Call-ID = 479

                                  09:54:32.115 SB.CCM  :  Org Acct = T01    Dst Acct = T03

                                  09:54:32.115 SB.CCM  :  Org Port ID = SipTrunk 0/0   Dst Port ID = unknown 0/0

                                  09:54:32.115 SB.CCM  :  SDP Transaction = CallID: 479

                                  09:54:32.115 SB.CCM  :  SDP Offer = 0x75201310, (88.88.88.88:44666)

                                  09:54:32.116 SB.CCM isMappable: Call Connection Type is RTP_TO_RTP

                                  09:54:32.116 SB.CCM handleRtpToRtp: Modifying SDP Offer

                                  09:54:32.116 SB.CCM translateOffer: offer codec list: PCMU G729

                                  09:54:32.117 SB.CCM translateOffer: revised offer codec list: PCMU

                                  09:54:32.117 SB.CCM translateOffer: codec list after answerer: PCMU

                                  09:54:32.118 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through

                                  09:54:32.118 SB.CCM translateOffer: success

                                  09:54:32.118 MEDIA.MANAGER Allocating media port.

                                  09:54:32.119 MEDIA.MANAGER getSubstitutePort: No matching callIdMap entry found for call 479

                                  09:54:32.119 MEDIA.MANAGER Call ID map : Added new entry : call ID 479 : session -3249275585INIP488.88.88.88 : version 951176 : index 1712

                                  09:54:32.119 MEDIA.MANAGER New media entry : type(0), callID(479), sessionID(-3249275585INIP488.88.88.88), original IP(88.88.88.88) ports(44666-44667), substitute IP(::) ports(11712-11713), RtpChannel(NULL), connection(0x0x75313710), sdpOverride(0), me(0x0x75202310). No RtpChannel

                                  09:54:32.119 SB.CALL 479 Idle                 Call sent from T01 to T03 (13237360311)

                                  09:54:32.120 SB.CALL 479 State change      >> Idle->Delivering

                                  09:54:32.120 TA.T01 01 TAInboundCall        CallResp event accepted

                                  09:54:32.120 TA.T01 01 State change      >> TAInboundCall->TAConnectWaitIn (TAS_Calling)

                                  09:54:32.120 TA.T03 91 State change      >> TAIdle->TAOutGoing (TAS_Delivering)

                                  09:54:32.121 TM.T03 91 SipTM_Idle           State change      >> SipTM_Idle->Delivering

                                  09:54:32.121 TM.T03 91 Delivering           Applying E.164 settings to called party number (13237360311)

                                  09:54:32.121 TM.T03 91 Delivering           Skipping E.164 conversion due to voice international-prefix setting

                                  09:54:32.121 TM.T03 91 Delivering           Applying E.164 settings to calling party number (13235551212)

                                  09:54:32.121 TM.T03 91 Delivering           From user grammar setting is: domestic

                                  09:54:32.122 TM.T03 91 Delivering           Skipping E.164 conversion due to From user grammar setting

                                  09:54:32.122 TM.T03 91 Looking up source address for destination 192.168.33.1

                                  09:54:32.122 TM.T03 91 call-leg (0x0x63161f60) -> src: 76.10.76.10: 5060  dst: 192.168.33.1 : 5060

                                  09:54:32.123 TM.T03 91 SDP DPI call ID 479 : No media bin.

                                  09:54:32.123 TM.T03 91 Processing new SDP entries.

                                  09:54:32.123 TM.T03 91 Checking for internal Media Gateway IP Address

                                  09:54:32.123 TM.T03 91 RTP Channel is NULL, Media Gateway must not be involved in call

                                  09:54:32.124 TM.T03 91 Undo of previous operation not required (RTP NAT Entry for 88.88.88.88:44666 not found)

                                  09:54:32.124 TM.T03 91 Checking for internal Media Gateway IP Address

                                  09:54:32.124 TM.T03 91 Given RTP Channel is null, checking for hairpinned RTP Channel

                                  09:54:32.124 TM.T03 91 RTP Channel is NULL, Media Gateway must not be involved in call

                                  09:54:32.124 TM.T03 91 Checking need for firewall traversal

                                  09:54:32.124 TM.T03 91 Testing firewall policies

                                  09:54:32.125 TM.T03 91 NAT not required, no need for firewall traversal here

                                  09:54:32.126 TM.T03 91 Delivering           call-leg -> Inviting

                                  09:54:32.127 TM.T03 91 Delivering           sent: INVITE

                                  09:54:32.127 SB.CALL 479 Delivering           Called the deliverResponse routine from Delivering

                                  09:54:32.127 SB.CALL 479 Delivering           DeliverResponse(accept) sent from T03 to T01

                                  09:54:32.128 TA.T01 01 TAConnectWaitIn      deliverResponse event accepted

                                  09:54:32.128 TA.T01 01 TAConnectWaitIn      ERROR! deliverResponse ignored

                                  09:54:32 SB.CallStructObserver 479 Created

                                  09:54:32 SB.CallStructObserver 479 <-> SDvpmc201-8d7cb69aee75599fcfc00a6d83e883a0-523g533

                                  09:54:35.128 TM.T03 91 INVITE rollover timeout

                                  09:54:35.128 TM.T03 91 Delivering           Sip_CreateCallLegNextServer with default validator

                                  09:54:35.128 TM.T03 91 Delivering           State change      >> Delivering->SipTM_Closing

                                  09:54:35.129 TM.T03 91 SipTM_Closing        sent: TA->Clear

                                  09:54:35.129 TM.T03 91 SipTM_Closing        call-leg -> Terminated

                                  09:54:35.129 TA.T03 91 TAOutGoing           rcvd: clear from TM

                                  09:54:35.129 TA.T03 91 State change      >> TAOutGoing->TATrunkClearing (TAS_Clearing)

                                  09:54:35.130 TM.T03 91 SipTM_Closing        tachg -> TATrunkClearing

                                  09:54:35.130 TM.T03 91 SipTM_Closing        State change      >> SipTM_Closing->SipTM_Terminated

                                  09:54:35.130 TM.T03 91 SipTM_Terminated     sent: TA->AppearanceOff

                                  09:54:35.130 TM.T03 91 SipTM_Terminated     State change      >> SipTM_Terminated->SipTM_Idle

                                  09:54:35.130 SB.CALL 479 Delivering           Called the clearCall routine

                                  09:54:35.131 SB.CALL 479 Delivering           SIP Proxy rejected call to 13237360311 for survivability - no matching Proxy user

                                  09:54:35.131 SB.CALL 479 Delivering           No available resources on call from T01 to T03 (last attempt)

                                  09:54:35.131 SB.CALL 479 State change      >> Delivering->Clearing

                                  09:54:35.131 TA.T03 91 TATrunkClearing      rcvd: appearance off from TM

                                  09:54:35.131 TA.T03 91 State change      >> TATrunkClearing->TAClearingComplete (TAS_Clearing)

                                  09:54:35.132 TA.T03 91 TATrunkClearing      Processing an appearance OFF

                                  09:54:35.132 TA.T01 01 TAConnectWaitIn      ClearCall event accepted

                                  09:54:35.132 TA.T01 01 State change      >> TAConnectWaitIn->TAClearingComplete (TAS_Clearing)

                                  09:54:35.132 TM.T01 01 SipTM_Pending        tachg -> TAClearingComplete

                                  09:54:35.132 TM.T01 01 SipTM_Pending        State change      >> SipTM_Pending->SipTM_CallFail

                                  09:54:35.133 TM.T01 01 SipTM_CallFail       call-leg -> Disconnected

                                  09:54:35.134 TM.T01 01 SipTM_CallFail       CallLegStateChanged to Disconnected - TM change to closing state.

                                  09:54:35.134 TM.T01 01 SipTM_CallFail       State change      >> SipTM_CallFail->SipTM_Closing

                                  09:54:35.134 TM.T01 01 SipTM_Closing        sent: TA->Clear

                                  09:54:35.134 TM.T01 01 SipTM_CallFail       sent: 503

                                  09:54:35.134 TM.T01 01 SipTM_Closing        State change      >> SipTM_Closing->SipTM_Terminated

                                  09:54:35.135 TM.T01 01 SipTM_Terminated     sent: TA->AppearanceOff

                                  09:54:35.135 TM.T01 01 SipTM_Terminated     State change      >> SipTM_Terminated->SipTM_Idle

                                  09:54:35.135 SB.CALL 479 Clearing             Called the clearResponse routine

                                  09:54:35.135 SB.CALL 479 State change      >> Clearing->CallIdlePending

                                  09:54:35.136 SB.CCM release:

                                  09:54:35.136 SB.CCM  :  Call Struct 0x0x7520e610 :   Call-ID = 479

                                  09:54:35.136 SB.CCM  :  Org Acct = T01    Dst Acct = T03

                                  09:54:35.136 SB.CCM  :  Org Port ID = SipTrunk 0/0   Dst Port ID = SipTrunk 0/0.290

                                  09:54:35.136 SB.CCM  :  SDP Transaction = CallID: 479

                                  09:54:35.137 SB.CCM  :  SDP Offer = 0x75201310, (88.88.88.88:44666)

                                  09:54:35.137 SB.CCM release: Call Connection Type is RTP_TO_RTP

                                  09:54:35.137 SB.CALL 479 CallIdlePending      ClearResponse sent from T01 to T03

                                  09:54:35.137 TA.T01 01 TAClearingComplete   rcvd: clear from TM

                                  09:54:35.137 TA.T01 01 TAClearingComplete   rcvd: appearance off from TM

                                  09:54:35.137 TA.T01 01 TAClearingComplete   Clear Local Variables

                                  09:54:35.138 TA.T01 01 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

                                  09:54:35.138 TM.T01 01 SipTM_Idle           tachg -> TAIdle

                                  09:54:35.138 TA.T03 91 TAClearingComplete   clearResponse event accepted

                                  09:54:35.138 TA.T03 91 TAClearingComplete   Clear Local Variables

                                  09:54:35.138 TA.T03 91 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

                                  09:54:35.139 TM.T03 91 SipTM_Idle           tachg -> TAIdle

                                  09:54:35 SB.CallStructObserver 479 Finalized

                                  2019.04.11 09:54:36 SMDR 479        04/11/2019 09:54:32      0.0 0    E  00/00 Bernard, St 13235551212      00/00 T03             13237360311     0 N  no debug voice verbose

                                  RDLINC#

                              • Re: Inbound calls not making it to softswitch
                                g-man New Member

                                Is there any way to make this work directly between the softswitch and SIP Provider and bypass the adtran for SIP? I can register and make outbound calls going directly to the provider, just need a way to get the calls to the softswitch. Basically just use it as a Router

                                  • Re: Inbound calls not making it to softswitch
                                    jayh Hall_of_Fame

                                    g-man wrote:

                                     

                                    Is there any way to make this work directly between the softswitch and SIP Provider and bypass the adtran for SIP? I can register and make outbound calls going directly to the provider, just need a way to get the calls to the softswitch. Basically just use it as a Router

                                    You could put a switch on the public side ahead of the Adtran and configure the softswitch to be directly on the Internet. Make sure that you have security very well locked down on the softswitch. It also makes calls between the softswitch and the PRI take a sub-optimal path. I'd use the Adtran even if it means shelling out for the license or going back to older firmware before Adtran crippled it. More flexible and secure.

                                     

                                    As far as registering the softswitch to the Adtran, you can do this but there is really no need to if it's directly on the LAN connected to the Adtran. Just reference it by IP address in the trunk.