13 Replies Latest reply on Jul 20, 2017 1:17 PM by brucekeller

    908 fails to receive calls

    astrosean New Member

      Hello everyone, 

       

      I'm new to the TDM world even though I've worked in the SIP world just for a few years. 

      The company I work for uses a lot of Adtran devices (900 series at customer premise)

       

      We have an Adtran 908 at a customer premise.  This device is connected to our network via IP via vpn  (No NAT).  I have access to both SSH and web gui.

      It was previously configured and working but hasn't been used in 6 months or more.  I can see that the device is set to peer to the sip server and shows ready. When I call a number that is directed to it,  the adtran receives the call, but seems to redirect it back to our Sonus.

       

      Suggestions?  I'm unsure what information is needed as I'm so green to Adtran.  

        • Re: 908 fails to receive calls
          jayh Hall_of_Fame

          astrosean wrote:

           

          It was previously configured and working but hasn't been used in 6 months or more. I can see that the device is set to peer to the sip server and shows ready. When I call a number that is directed to it, the adtran receives the call, but seems to redirect it back to our Sonus.

           

          Where is the call supposed to go? Is there a PRI connected to it? A SIP PBX? Individual analog users? It could be that the call is reaching the customer PBX and that is hairpinning the call back. Can the customer place outbound calls OK?

           

          A sanitized configuration (remove passwords and public IP info) and the results of "debug voice verbose" when a call is placed would be useful.

            • Re: 908 fails to receive calls
              astrosean New Member

              Here you go..

               

              Call is supposed to go to a FXS port (analog user).

              Unsure about outbound calls.  I have a call into the on premise to test outbound calls.  Awaiting response from them.

               

              Here is a sanitized copy of the config.

               

              The number of concern is 7273025555

               

              !

              !

              !

              hostname "helpme"

              enable password boom12345

              !

              clock timezone -5-Eastern-Time

              !

              statistics rate-interval 30

              !

              ip subnet-zero

              ip classless

              ip default-gateway 10.0.250.1

              ip routing

              !

              !

               

               

              ip domain-name "sometelecom.net"

              ip name-server 8.8.8.8 4.2.2.3

              !

              !

              no auto-config

              !

              event-history on

              no logging forwarding

              no logging email

              !

              no service password-encryption

              !

              username "admin" password "adminpass123abc"

              username "mike" password "boom12345"

              username "adtran" password "flylikeabird"

              !

              !

              no ip firewall alg msn

              no ip firewall alg mszone

              no ip firewall alg h323

              !

              !

              !

              !

              !

              no dot11ap access-point-control

              !

              !

              !

              !

              !

              ip dhcp-server pool "Test"

              !

              !

              !

              !

              !

              !

              qos map ConfigWizardQoSMap 20

                match dscp 46

                priority unlimited

              !

              !

              !

              !

              interface eth 0/1

                description HelpMe

                ip address  10.0.250.2  255.255.255.0

                media-gateway ip primary

                no shutdown

              !

              !

              !

              !

              interface t1 0/1

                description PRI Port

                shutdown

              !

              interface t1 0/2

                description helpme

                tdm-group 1 timeslots 1-24 speed 64

                no shutdown

              !

              !

              interface pri 1

                description helpme

                isdn name-delivery setup

                connect t1 0/2 tdm-group 1

                role network b-channel-restarts enable

                no shutdown

              !

              !

              interface fxs 0/1

                description "7273024444"

                no shutdown

              !

              interface fxs 0/2

                description "7273025555"

                no shutdown

              !

              interface fxs 0/3

                shutdown

              !

              interface fxs 0/4

                shutdown

              !

              interface fxs 0/5

                impedance 600r

                shutdown

              !

              interface fxs 0/6

                impedance 600r

                shutdown

              !

              interface fxs 0/7

                impedance 600r

                shutdown

              !

              interface fxs 0/8

                impedance 600r

                shutdown

              !

              !

              isdn-group 1

                connect pri 1

              !

              !

              !

              !

              timing-source internal

              !

              timing-source internal secondary

              !

              !

              !

              !

              !

              !

              !

              ip route 0.0.0.0 0.0.0.0 10.0.250.1

              ip route 0.0.0.0 0.0.0.0 71.41.86.201

              ip route 0.0.0.0 0.0.0.0 96.252.180.1

              !

              no ip tftp server

              no ip tftp server overwrite

              ip http server

              ip http session-timeout 1200

              ip http session-limit 10

              ip http secure-server

              no ip snmp agent

              no ip ftp server

              no ip scp server

              ip sntp server

              !

              !

              !

              !

              !

              !

              ip sip

              ip sip udp 5060

              no ip sip tcp

              !

              !

              !

              voice feature-mode network

              voice forward-mode network

              voice call-appearance-mode single

              !

              !

              !

              !

              !

              !

              !

              voice dial-plan 1 local NXX-NXX-XXXX

              voice dial-plan 3 long-distance 1-NXX-NXX-XXXX

              voice dial-plan 4 toll-free 1-800-NXX-XXXX

              voice dial-plan 5 toll-free 1-877-NXX-XXXX

              voice dial-plan 6 toll-free 1-866-NXX-XXXX

              voice dial-plan 7 toll-free 1-855-NXX-XXXX

              voice dial-plan 8 always-permitted 511

              voice dial-plan 9 always-permitted 411

              voice dial-plan 10 always-permitted 1-411

              voice dial-plan 11 always-permitted 555-1212

              voice dial-plan 12 international 011-XXXXXXXXXX

              voice dial-plan 13 international 011-XXXXXXXXXXX

              voice dial-plan 14 international 011-XXXXXXXXXXXX

              voice dial-plan 15 international 011-XXXXXXXXXXXXX

              voice dial-plan 16 international 011-XXXXXXXXXXXXXX

              voice dial-plan 17 international 011-XXXXXXXXXXXXXXX

              voice dial-plan 18 international 011-XXXXXXXXXXXXXXXX

              voice dial-plan 19 local 611

              !

              !

              !

              !

              voice class-of-service flylikeabirdTrk

                default-level

                billing-codes

                call-privilege extensions

                call-privilege international

                call-privilege local

                call-privilege long-distance

                call-privilege operator-assisted

                call-privilege specify-carrier

                call-privilege toll-free

                call-privilege 900-number

              !

              voice codec-list whirlpool

                default

                codec g711ulaw

              !

              !

              !

              voice trunk T01 type sip

                description "sonus"

                sip-server primary 192.192.192.99

              !

              voice trunk T02 type isdn

                description "ISDN"

                resource-selection linear descending

                caller-id-override number-inbound 7275737663 if-no-cpn

                connect isdn-group 1

                alc

                modem-passthrough

                t38

                plc

                rtp delay-mode adaptive

              !

              !

              voice grouped-trunk "SIP TRUNK"

                trunk T01

                accept $ cost 0

                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 911 cost 0

                reject NXX-976-XXXX

                reject 1-900-NXX-XXXX

                reject 1-976-NXX-XXXX

              !

              !

              voice grouped-trunk "ISDN TRUNK"

                trunk T02

                accept $ cost 10

              !

              !

              voice user 7273024444

                connect fxs 0/1

                cos "flylikeabirdTrk"

                first-name "Alarm"

                last-name "Line"

                password "1234"

                description "7273024444"

                sip-identity 7273024444 T01

                sip-authentication password "1234"

                no nls

                no echo-cancellation

                codec-group whirlpool

              !

              !

              voice user 7273025555

                connect fxs 0/2

                cos "flylikeabirdTrk"

                first-name "conference"

                last-name "room"

                password "1234"

                sip-identity 7273025555 T01

                sip-authentication password "1234"

                no echo-cancellation

                codec-group whirlpool

              !

              !

              !

              !

              !

              !

              !

              !

              !

              !

              no ip sip registrar authenticate

              ip sip registrar default-expires 5000

              ip sip registrar min-expires 3600

              ip sip registrar max-expires 5000

              !

              !

              !

              !

              !

              !

              !

              !

              !

              !

              !

              ip rtp udp 16384

              !

              ip sdp grammar hold rfc3264

              !

              !

              line con 0

                login

              !

              line telnet 0 4

                login

                password boom12345

                shutdown

              line ssh 0 4

                login local-userlist

                no shutdown

              !

              sntp server 216.182.242.7 version 3

              sntp wait-time 500

              !

              !

              !

              !

              end

               

               

               

               

              Tried capturing voice verbose, but nothing happened during the failed call.  However here are the results of SIP all.

               

              12:16:12.467 SIP. MSG OPTIONS REQ RX unknown unknown

              OPTIONS sip:10.0.250.2:5060 SIP/2.0

              From: <sip:192.192.192.99>;tag=gK004cb43d

              To: <sip:10.0.250.2>

              Call-ID: 137742052_100523466@192.192.192.99

              CSeq: 700098 OPTIONS

              Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B163be93d364f02d4

              Max-Forwards: 1

              Contact: <sip:192.192.192.99:5060>

              Content-Length: 0

              12:16:12.468 SIP. MSGSUM OPTIONS REQ RX unknown unknown sip:10.0.250.2:5060

              12:16:12.469 SIP.TDU Transaction accepted by SIP Server

              12:16:12.470 SIP.TDU Incoming message received from 192.192.192.99:5060 via UDP

              12:16:12.472 SIP. MSG OPTIONS RSP TX unknown unknown

              SIP/2.0 200 OK

              From: <sip:192.192.192.99>;tag=gK004cb43d

              To: <sip:10.0.250.2>;tag=24c7c28-7f000001-13c4-1915-fb46dcb6-1915

              Call-ID: 137742052_100523466@192.192.192.99

              CSeq: 700098 OPTIONS

              Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B163be93d364f02d4

              Supported: 100rel,replaces

              Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

              User-Agent: ADTRAN_Total_Access_908_1st_Gen/A4.11.00.E

              Content-Length: 0

              12:16:12.473 SIP. MSGSUM OPTIONS RSP TX unknown unknown 200 OK

                • Re: 908 fails to receive calls
                  jayh Hall_of_Fame

                  If nothing happened during the debug voice verbose, it sounds like the call either never hit the TA900 or the TA900 isn't accepting its SIP.

                   

                  The SIP capture is an options message and doesn't look like it is necessarily related to your call.

                   

                  Does "debug sip stack messages" show an INVITE message?

                   

                  Also, while not related to this specific problem, your dial plan could use some tweaks. To avoid potential issues, you probably want to change the following to "local" instead of "always-permitted".

                   

                  voice dial-plan 8 always-permitted 511

                  voice dial-plan 9 always-permitted 411

                  voice dial-plan 10 always-permitted 1-411

                  voice dial-plan 11 always-permitted 555-1212

                   

                  always-permitted is for emergency calls.

                   

                  And you should have:

                   

                  voice dial-plan 2 always-permitted 911

                   

                  In addition, 1-844-NXX-XXXX and 1-833-NXX-XXXX are now valid toll-free patterns.

                    • Re: 908 fails to receive calls
                      astrosean New Member

                      Nice catch on those btw.  Thank you!

                       

                       

                      Here's the latest debug capture

                       

                      14:46:13.266 SIP.STACK MSG     Rx: UDP src=192.192.192.99:5060 dst=10.0.250.2:5060

                      14:46:13.267 SIP.STACK MSG         OPTIONS sip:10.0.250.2:5060 SIP/2.0

                      14:46:13.267 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1be13cc96d3ec07d

                      14:46:13.268 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003e22df

                      14:46:13.268 SIP.STACK MSG         To: <sip:10.0.250.2>

                      14:46:13.269 SIP.STACK MSG         Call-ID: 137793055_129880511@192.192.192.99

                      14:46:13.269 SIP.STACK MSG         CSeq: 62701 OPTIONS

                      14:46:13.270 SIP.STACK MSG         Max-Forwards: 1

                      14:46:13.270 SIP.STACK MSG         Contact: <sip:192.192.192.99:5060>

                      14:46:13.271 SIP.STACK MSG         Content-Length: 0

                      14:46:13.272 SIP.STACK MSG    

                      14:46:13.282 SIP.STACK MSG     Tx: UDP src=10.0.250.2:5060 dst=192.192.192.99:5060

                      14:46:13.283 SIP.STACK MSG         SIP/2.0 200 OK

                      14:46:13.283 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003e22df

                      14:46:13.284 SIP.STACK MSG         To: <sip:10.0.250.2>;tag=24b7008-7f000001-13c4-3c3e-feb52672-3c3e

                      14:46:13.285 SIP.STACK MSG         Call-ID: 137793055_129880511@192.192.192.99

                      14:46:13.285 SIP.STACK MSG         CSeq: 62701 OPTIONS

                      14:46:13.286 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1be13cc96d3ec07d

                      14:46:13.286 SIP.STACK MSG         Supported: 100rel,replaces

                      14:46:13.287 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

                      14:46:13.287 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908_1st_Gen/A4.11.00.E

                      14:46:13.288 SIP.STACK MSG         Content-Length: 0

                      14:46:13.288 SIP.STACK MSG    

                      14:46:28.266 SIP.STACK MSG     Rx: UDP src=192.192.192.99:5060 dst=10.0.250.2:5060

                      14:46:28.266 SIP.STACK MSG         OPTIONS sip:10.0.250.2:5060 SIP/2.0

                      14:46:28.267 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1c5dc0d28390b748

                      14:46:28.267 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003e93d4

                      14:46:28.268 SIP.STACK MSG         To: <sip:10.0.250.2>

                      14:46:28.268 SIP.STACK MSG         Call-ID: 137793140_133644708@192.192.192.99

                      14:46:28.269 SIP.STACK MSG         CSeq: 612431 OPTIONS

                      14:46:28.269 SIP.STACK MSG         Max-Forwards: 1

                      14:46:28.270 SIP.STACK MSG         Contact: <sip:192.192.192.99:5060>

                      14:46:28.270 SIP.STACK MSG         Content-Length: 0

                      14:46:28.271 SIP.STACK MSG    

                      14:46:28.281 SIP.STACK MSG     Tx: UDP src=10.0.250.2:5060 dst=192.192.192.99:5060

                      14:46:28.282 SIP.STACK MSG         SIP/2.0 200 OK

                      14:46:28.282 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003e93d4

                      14:46:28.283 SIP.STACK MSG         To: <sip:10.0.250.2>;tag=24b71c0-7f000001-13c4-3c4d-a21f0c2d-3c4d

                      14:46:28.283 SIP.STACK MSG         Call-ID: 137793140_133644708@192.192.192.99

                      14:46:28.284 SIP.STACK MSG         CSeq: 612431 OPTIONS

                      14:46:28.284 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1c5dc0d28390b748

                      14:46:28.285 SIP.STACK MSG         Supported: 100rel,replaces

                      14:46:28.285 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

                      14:46:28.286 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908_1st_Gen/A4.11.00.E

                      14:46:28.286 SIP.STACK MSG         Content-Length: 0

                      14:46:28.287 SIP.STACK MSG    

                      14:46:43.266 SIP.STACK MSG     Rx: UDP src=192.192.192.99:5060 dst=10.0.250.2:5060

                      14:46:43.266 SIP.STACK MSG         OPTIONS sip:10.0.250.2:5060 SIP/2.0

                      14:46:43.267 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1cdce041137e0640

                      14:46:43.267 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003f0b30

                      14:46:43.268 SIP.STACK MSG         To: <sip:10.0.250.2>

                      14:46:43.268 SIP.STACK MSG         Call-ID: 137793225_125491704@192.192.192.99

                      14:46:43.269 SIP.STACK MSG         CSeq: 336093 OPTIONS

                      14:46:43.270 SIP.STACK MSG         Max-Forwards: 1

                      14:46:43.270 SIP.STACK MSG         Contact: <sip:192.192.192.99:5060>

                      14:46:43.271 SIP.STACK MSG         Content-Length: 0

                      14:46:43.271 SIP.STACK MSG    

                      14:46:43.282 SIP.STACK MSG     Tx: UDP src=10.0.250.2:5060 dst=192.192.192.99:5060

                      14:46:43.282 SIP.STACK MSG         SIP/2.0 200 OK

                      14:46:43.283 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003f0b30

                      14:46:43.283 SIP.STACK MSG         To: <sip:10.0.250.2>;tag=24b7378-7f000001-13c4-3c5c-fa1b81a9-3c5c

                      14:46:43.284 SIP.STACK MSG         Call-ID: 137793225_125491704@192.192.192.99

                      14:46:43.284 SIP.STACK MSG         CSeq: 336093 OPTIONS

                      14:46:43.285 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1cdce041137e0640

                      14:46:43.285 SIP.STACK MSG         Supported: 100rel,replaces

                      14:46:43.286 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

                      14:46:43.287 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908_1st_Gen/A4.11.00.E

                      14:46:43.287 SIP.STACK MSG         Content-Length: 0

                      14:46:43.288 SIP.STACK MSG    

                      14:46:58.272 SIP.STACK MSG     Rx: UDP src=192.192.192.99:5060 dst=10.0.250.2:5060

                      14:46:58.272 SIP.STACK MSG         OPTIONS sip:10.0.250.2:5060 SIP/2.0

                      14:46:58.273 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1d51e6fd2a5d811d

                      14:46:58.274 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003f7657

                      14:46:58.274 SIP.STACK MSG         To: <sip:10.0.250.2>

                      14:46:58.275 SIP.STACK MSG         Call-ID: 137793310_133810858@192.192.192.99

                      14:46:58.275 SIP.STACK MSG         CSeq: 222758 OPTIONS

                      14:46:58.276 SIP.STACK MSG         Max-Forwards: 1

                      14:46:58.276 SIP.STACK MSG         Contact: <sip:192.192.192.99:5060>

                      14:46:58.277 SIP.STACK MSG         Content-Length: 0

                      14:46:58.277 SIP.STACK MSG    

                      14:46:58.288 SIP.STACK MSG     Tx: UDP src=10.0.250.2:5060 dst=192.192.192.99:5060

                      14:46:58.288 SIP.STACK MSG         SIP/2.0 200 OK

                      14:46:58.289 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003f7657

                      14:46:58.289 SIP.STACK MSG         To: <sip:10.0.250.2>;tag=24b7530-7f000001-13c4-3c6b-bcb55878-3c6b

                      14:46:58.290 SIP.STACK MSG         Call-ID: 137793310_133810858@192.192.192.99

                      14:46:58.290 SIP.STACK MSG         CSeq: 222758 OPTIONS

                      14:46:58.291 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1d51e6fd2a5d811d

                      14:46:58.292 SIP.STACK MSG         Supported: 100rel,replaces

                      14:46:58.292 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

                      14:46:58.293 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908_1st_Gen/A4.11.00.E

                      14:46:58.293 SIP.STACK MSG         Content-Length: 0

                      14:46:58.294 SIP.STACK MSG    

                      14:47:13.273 SIP.STACK MSG     Rx: UDP src=192.192.192.99:5060 dst=10.0.250.2:5060

                      14:47:13.274 SIP.STACK MSG         OPTIONS sip:10.0.250.2:5060 SIP/2.0

                      14:47:13.275 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1dc508aa00513a58

                      14:47:13.275 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003fe2e0

                      14:47:13.276 SIP.STACK MSG         To: <sip:10.0.250.2>

                      14:47:13.276 SIP.STACK MSG         Call-ID: 137793395_125572646@192.192.192.99

                      14:47:13.277 SIP.STACK MSG         CSeq: 604056 OPTIONS

                      14:47:13.277 SIP.STACK MSG         Max-Forwards: 1

                      14:47:13.278 SIP.STACK MSG         Contact: <sip:192.192.192.99:5060>

                      14:47:13.278 SIP.STACK MSG         Content-Length: 0

                      14:47:13.279 SIP.STACK MSG    

                      14:47:13.289 SIP.STACK MSG     Tx: UDP src=10.0.250.2:5060 dst=192.192.192.99:5060

                      14:47:13.290 SIP.STACK MSG         SIP/2.0 200 OK

                      14:47:13.291 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK003fe2e0

                      14:47:13.291 SIP.STACK MSG         To: <sip:10.0.250.2>;tag=24b76e8-7f000001-13c4-3c7a-a420a5d8-3c7a

                      14:47:13.292 SIP.STACK MSG         Call-ID: 137793395_125572646@192.192.192.99

                      14:47:13.292 SIP.STACK MSG         CSeq: 604056 OPTIONS

                      14:47:13.293 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1dc508aa00513a58

                      14:47:13.293 SIP.STACK MSG         Supported: 100rel,replaces

                      14:47:13.294 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

                      14:47:13.294 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908_1st_Gen/A4.11.00.E

                      14:47:13.295 SIP.STACK MSG         Content-Length: 0

                      14:47:13.295 SIP.STACK MSG    

                      14:47:28.274 SIP.STACK MSG     Rx: UDP src=192.192.192.99:5060 dst=10.0.250.2:5060

                      14:47:28.275 SIP.STACK MSG         OPTIONS sip:10.0.250.2:5060 SIP/2.0

                      14:47:28.275 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1e3da2bd8ab4b8bd

                      14:47:28.276 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK00404d1f

                      14:47:28.277 SIP.STACK MSG         To: <sip:10.0.250.2>

                      14:47:28.277 SIP.STACK MSG         Call-ID: 137793480_116376952@192.192.192.99

                      14:47:28.278 SIP.STACK MSG         CSeq: 952914 OPTIONS

                      14:47:28.278 SIP.STACK MSG         Max-Forwards: 1

                      14:47:28.279 SIP.STACK MSG         Contact: <sip:192.192.192.99:5060>

                      14:47:28.279 SIP.STACK MSG         Content-Length: 0

                      14:47:28.280 SIP.STACK MSG    

                      14:47:28.290 SIP.STACK MSG     Tx: UDP src=10.0.250.2:5060 dst=192.192.192.99:5060

                      14:47:28.291 SIP.STACK MSG         SIP/2.0 200 OK

                      14:47:28.291 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK00404d1f

                      14:47:28.292 SIP.STACK MSG         To: <sip:10.0.250.2>;tag=24b78a0-7f000001-13c4-3c89-eed3bfe3-3c89

                      14:47:28.292 SIP.STACK MSG         Call-ID: 137793480_116376952@192.192.192.99

                      14:47:28.293 SIP.STACK MSG         CSeq: 952914 OPTIONS

                      14:47:28.293 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1e3da2bd8ab4b8bd

                      14:47:28.294 SIP.STACK MSG         Supported: 100rel,replaces

                      14:47:28.295 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

                      14:47:28.295 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908_1st_Gen/A4.11.00.E

                      14:47:28.296 SIP.STACK MSG         Content-Length: 0

                      14:47:28.296 SIP.STACK MSG    

                      14:47:43.272 SIP.STACK MSG     Rx: UDP src=192.192.192.99:5060 dst=10.0.250.2:5060

                      14:47:43.272 SIP.STACK MSG         OPTIONS sip:10.0.250.2:5060 SIP/2.0

                      14:47:43.273 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1eaef3098d28c1c6

                      14:47:43.273 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK0040b8bb

                      14:47:43.274 SIP.STACK MSG         To: <sip:10.0.250.2>

                      14:47:43.275 SIP.STACK MSG         Call-ID: 137793565_129486139@192.192.192.99

                      14:47:43.275 SIP.STACK MSG         CSeq: 790427 OPTIONS

                      14:47:43.276 SIP.STACK MSG         Max-Forwards: 1

                      14:47:43.276 SIP.STACK MSG         Contact: <sip:192.192.192.99:5060>

                      14:47:43.277 SIP.STACK MSG         Content-Length: 0

                      14:47:43.277 SIP.STACK MSG    

                      14:47:43.288 SIP.STACK MSG     Tx: UDP src=10.0.250.2:5060 dst=192.192.192.99:5060

                      14:47:43.288 SIP.STACK MSG         SIP/2.0 200 OK

                      14:47:43.289 SIP.STACK MSG         From: <sip:192.192.192.99>;tag=gK0040b8bb

                      14:47:43.289 SIP.STACK MSG         To: <sip:10.0.250.2>;tag=24b7a58-7f000001-13c4-3c98-d4678047-3c98

                      14:47:43.290 SIP.STACK MSG         Call-ID: 137793565_129486139@192.192.192.99

                      14:47:43.290 SIP.STACK MSG         CSeq: 790427 OPTIONS

                      14:47:43.291 SIP.STACK MSG         Via: SIP/2.0/UDP 192.192.192.99:5060;branch=z9hG4bK00B1eaef3098d28c1c6

                      14:47:43.291 SIP.STACK MSG         Supported: 100rel,replaces

                      14:47:43.292 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

                      14:47:43.292 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908_1st_Gen/A4.11.00.E

                      14:47:43.293 SIP.STACK MSG         Content-Length: 0

                      14:47:43.293 SIP.STACK MSG    

                        • Re: 908 fails to receive calls
                          jayh Hall_of_Fame

                          Your issue is with the SIP server initiating the call and not with the analog FXS station.

                           

                          Generally, a call is initiated with a SIP INVITE message having the call details such as calling and called number, IP and port for the RTP session (SDP), and other data related to the call.

                           

                          Your provider, however, first sends an OPTIONS message to determine if your device is alive and ready to accept a call. It's kind of like asking "Are you there?" before delivering a message. OPTIONS can also query such things as SDP capabilities and the like, but this doesn't seem to be the case here. If the sender gets a response that indicates the call will fail, or gets no response at all, then no INVITE is ever sent.

                           

                          Looking at the debug, you receive the OPTIONS message and immediately respond with an OK stating your capabilities including that you can accept an INVITE.

                           

                          Apparently, the SIP server at 192.192.192.99 never receives your reply. Therefore it never sends the INVITE. Seeing as the OPTIONS messages are received every 15 seconds exactly, my guess is that the sending server is not receiving your replies and is set to retry three times at 15-second intervals, then give up.

                           

                          This could be an IP routing issue, a firewall rule somewhere, or simply broken behavior on the part SIP sender. Further examination of your configuration indicates something odd. You have:

                           

                          !

                          ip route 0.0.0.0 0.0.0.0 10.0.250.1

                          ip route 0.0.0.0 0.0.0.0 71.41.86.201

                          ip route 0.0.0.0 0.0.0.0 96.252.180.1

                          !

                           

                          Yet the only connected IP interface is eth 0/1 which is in the 10.0.250.0/24 space. Therefore your second two default routes should be removed.

                           

                          However, 10.0.250.0/24 is a private address. You say there's a VPN between it and your network with no NAT. Is the SIP server "sonus" also on your network with no NAT, or does it traverse an external NAT to go to the Internet to reach "sonus"? You will want to avoid NAT between the Adtran TA900 and the SIP server.

                           

                          At any rate, if the response to the OPTIONS message isn't received and processed, then the INVITE will also almost certainly fail.

                           

                          Troubleshoot the path from your TA908 back to SIP server "sonus" and you'll probably find the issue.

                            • Re: 908 fails to receive calls
                              astrosean New Member

                              I've removed the additional default routes and I've verified two way connectivity between the 908 and the Sonus.  Also, I've tripled checked the vpn config, and the routing is without NAT.   (When we deploy a customer device, we reserve the IP space used in the private space within our network and just route the packets.   We mainly did this before net neutrality was a thing, and now its even more important. )

                               

                              BTW, is there a guide or a tree to lookup all the CLI commands for the 900 series and how they are used?

                                • Re: 908 fails to receive calls
                                  jayh Hall_of_Fame

                                  astrosean wrote:

                                   

                                  I've removed the additional default routes and I've verified two way connectivity between the 908 and the Sonus. Also, I've tripled checked the vpn config, and the routing is without NAT. (When we deploy a customer device, we reserve the IP space used in the private space within our network and just route the packets. We mainly did this before net neutrality was a thing, and now its even more important. )

                                   

                                  OK, next step would be to determine why the Sonus is unhappy with (or ignoring) the 200OK responses to the OPTIONS messages, or get it to not do the OPTIONS thing and just send an INVITE.

                                   

                                  The problem seems to be that the Sonus doesn't see (or doesn't like) the response to its OPTIONS query and thus never sets up the call.

                                   

                                  astrosean wrote:

                                   

                                  BTW, is there a guide or a tree to lookup all the CLI commands for the 900 series and how they are used?

                                   

                                  Oh, you must want "The Bible".  :-)  AOS Version R12.3 Command Reference Guide

                                    • Re: 908 fails to receive calls
                                      astrosean New Member

                                      Yep, already working withe Sonus engineer and explained to him that the invite doesn't seem to be sent.   

                                       

                                      For the Bible, have things changed much from v11 and v12?  Looks like the latest this 908 supports is v11.

                                        • Re: 908 fails to receive calls
                                          jayh Hall_of_Fame

                                          astrosean wrote:

                                           

                                          Yep, already working withe Sonus engineer and explained to him that the invite doesn't seem to be sent.

                                           

                                          For the Bible, have things changed much from v11 and v12? Looks like the latest this 908 supports is v11.

                                          Most of the commands are the same. There are some new ones that are only in R12. A couple of years ago they tweaked the use of the "ip" keyword to differentiate between IPv4 and IPv6. Mostly, the "ip" was removed from commands that would apply equally to v4 and v6 and added where the specific version needed to be specified. This was mostly within the various flavors of R10.

                                           

                                          When you upgrade, on the first reboot the configuration is parsed and tweaked as needed to be compatible with the new version. This works really well so no issues. However, if you ever need to roll back to an older version and the configuration was automatically modified, things will break. An archive utility like RANCID is your friend.

                                           

                                          I also recommend that you archive any Adtran documentation and firmware that you use locally as they have become pretty aggressive about removing older versions from the website.

                                            • Re: 908 fails to receive calls
                                              astrosean New Member

                                              Understood thank you!

                                               

                                              Well, still no luck on my side.

                                               

                                              Is there any kind of step by step tutorial to configure one of these 900 series from scratch as an peering ATA (no registration)?.   I feel as if I'm missing something and it'd do me some good to work with this as much as I can.

                                                • Re: 908 fails to receive calls
                                                  jayh Hall_of_Fame

                                                  astrosean wrote:

                                                   

                                                  Is there any kind of step by step tutorial to configure one of these 900 series from scratch as an peering ATA (no registration)?. I feel as if I'm missing something and it'd do me some good to work with this as much as I can.

                                                  Not per se that I'm aware of. Basic idea, set up a voice trunk type SIP with the other side as the SIP server (ip or hostname). Don't configure registration. Do the same on the other side with the ip of the TA900. You can do two of them back-to-back or set one up to peer with your SBC.

                                                   

                                                  Then use the grouped-trunk to configure the digit patterns to send to the other side.

                                                   

                                                  For more than one endpoint, build multiple trunks.

                                                   

                                                  The GUI is easier initially to do much of this, but the CLI is more powerful. You can use the GUI to build a simple configuration and then use the CLI to see what was done and modify it. If you're going to be working with these devices a lot, learn to do everything in the CLI. Learning curve isn't that steep.

                                                    • Re: 908 fails to receive calls
                                                      astrosean New Member

                                                      Just wanted to let you know we used a different sip server (our legacy one) and everything worked fine.  Just updated the voip trunk to reflect the different ip, applied and rebooted and voila, magic.   Go figure. 

                                                       

                                                      Thanks so much for helping a new guy.

                                                       

                                                      Now that the fire is out, going to put one on a bench and set it up from scratch from the CLI.  

                                    • Re: 908 fails to receive calls
                                      brucekeller New Member

                                      I'd probably edit that post with the full config file unless those passwords were already preedited before you pasted / you don't care.