8 Replies Latest reply on Nov 5, 2013 6:32 AM by matt

    Voice Loopback

    touristsis Visitor

      I've just learn that Adtran can do voice loopback - actually making calls for troubleshooting purpose.  Really cool concept, but I cannot get it to work to save my life.  Here is what I really don't understand.  If I already have a sip trunk up and running on the system.  Do I have to register the loopback again? What if I don't know the password, it's *** out?  What if my SIP provider only allow me to register once?  Does that mean I can't use that feature for testing.  It would be really cool if I can get this feature to work.  Any help is appreciated.


      Is there a way for me to tell it to use the existing trunk registration?

        • Re: Voice Loopback
          Employee

          Here is a link to our AOS voice loopback account guide: Configuring the AOS Voice Loopback Account .  You should be able to place calls from the loopback account without registering it to your SIP provider. Just make sure and change the ANI of calls originating from the loopback account with the caller-id number command under the loopback account to a number that has been registered already.  If that does not work and you can paste in a copy of a show run voice with any public IPs, phone numbers, passwords, or other sensitive information removed, I would happy to take a look and help troubleshoot.

           

          Thanks,

          Matt

            • Re: Voice Loopback
              touristsis Visitor

              Hi Matt,

               

              I've tried it.  When I run a debug voice switch switchboard, this is what I'm getting.

               

              12:47:06.795 SB.CALL 15 Idle                 Called the call routine with 9512695XXXX

              12:47:06 SB.TGMgr For dialed number 9512695XXXX6, against template 9-NXX-NXX-XXXX, on TrunkGroup BESTLINE, the score is 2000

              12:47:06.795 SB.CALL 15 Idle                 No LOCAL station matched dialed number (9512XXXX)

              12:47:06.796 SB.CCM isMappable:

              12:47:06.796 SB.CCM  :  Call Struct 0x4134010 :   Call-ID = 15

              12:47:06.796 SB.CCM  :  Org Acct = 115    Dst Acct = T06

              12:47:06.796 SB.CCM  :  Org Port ID = 0/0.0   Dst Port ID = 0/0.0

              12:47:06.796 SB.CCM  :  Org TID =    Dst TID =

              12:47:06.796 SB.CCM  :  SDP Transaction = CallID: 15

              12:47:06.797 SB.CCM  :  SDP Offer = 0x03d8b710, (127.0.0.1:10048)

              12:47:06.797 SB.CCM isMappable: Call Connection Type is RTP_TO_RTP

              12:47:06.797 SB.CCM handleRtpToRtp: Modifying SDP Offer

              12:47:06.798 SB.CCM translateOffer: offer codec list: G722 PCMU PCMA G729

              12:47:06.798 SB.CCM translateOffer: revised offer codec list: G722 PCMU PCMA G729

              12:47:06.799 SB.CCM translateOffer: codec list after answerer: PCMU G729

              12:47:06.800 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through

              12:47:06.800 SB.CCM translateOffer: success

              12:47:06.801 SB.CALL 15 Idle                 Call sent from 115 to T06 (9512695XXXX)

              12:47:06.801 SB.CALL 15 State change      >> Idle->Delivering

              12:47:06.808 SB.CALL 15 Delivering           Called the deliverResponse routine from Delivering

              12:47:06.808 SB.CALL 15 Delivering           DeliverResponse(accept) sent from T06 to 115

              12:47:06 SB.CallStructObserver 15 Created

              12:47:06.899 SB.CCM isResponseMappable:

              12:47:06.899 SB.CCM  :  Call Struct 0x4134010 :   Call-ID = 15

              12:47:06.900 SB.CCM  :  Org Acct = 115    Dst Acct = T06

              12:47:06.900 SB.CCM  :  Org Port ID = 0/0.0   Dst Port ID = 0/0.65

              12:47:06.900 SB.CCM  :  Org TID =    Dst TID =

              12:47:06.900 SB.CCM  :  SDP Transaction = CallID: 15

              12:47:06.900 SB.CCM  :  SDP Offer = 0x03d8b710, (127.0.0.1:10048)

              12:47:06.900 SB.CCM  :  SDP Answer = 0x03d8ab10, (216.16.200.237:38672)

              12:47:06.901 SB.CCM isResponseMappable: reversing call connection type to compensate for event originator direction

              12:47:06.901 SB.CCM isResponseMappable: Call Connection Type is RTP_TO_RTP

              12:47:06.901 SB.CCM handleRtpToRtp: Modifying SDP Answer

              12:47:06.901 SB.CCM translateAnswer: offer  codec list: G722 PCMU PCMA G729

              12:47:06.902 SB.CCM                : answer codec list: PCMU

              12:47:06.902 SB.CCM translateAnswer: CODEC transcoding is not required

              12:47:06.902 SB.CCM translateAnswer: skipping DTMF answer translation for (NTE 101->NTE 101) due to invalid resource(0x0) or unsupported CODEC, using (NTE 101)

              12:47:06.903 SB.CCM translateAnswer: success

              12:47:06.903 SB.CALL 15 Delivering           Sent preConnect from T06 to 115

              12:47:06.903 SB.CALL 15 State change      >> Delivering->PreConnecting

              12:47:06.904 SB.CCM connect:

              12:47:06.904 SB.CCM  :  Call Struct 0x4134010 :   Call-ID = 15

              12:47:06.904 SB.CCM  :  Org Acct = 115    Dst Acct = T06

              12:47:06.904 SB.CCM  :  Org Port ID = 0/0.0   Dst Port ID = 0/0.65

              12:47:06.904 SB.CCM  :  Org TID =    Dst TID =

              12:47:06.904 SB.CCM  :  SDP Transaction = CallID: 15

              12:47:06.905 SB.CCM  :  SDP Offer = 0x03d8b710, (127.0.0.1:10048)

              12:47:06.905 SB.CCM  :  SDP Answer = 0x03d8ab10, (216.16.200.237:38672)

              12:47:06.905 SB.CCM connect: Call Connection Type is RTP_TO_RTP

              12:47:06.906 SB.CCM connect: Connected RTP_TO_RTP via MCM

              12:47:06.906 SB.CCM handleMediaConnectionRtpChannel: No RTP channel to set up

              12:47:06.906 SB.CCM firewallConnectCall: Checking need for firewall traversal

              12:47:06.906 SB.CCM firewallConnectCall: Testing firewall policies

              12:47:06.907 SB.CCM firewallConnectCall: Creating firewall associations to connect 24.173.XX.XX:10048 to 216.16.XX.XX:38672

              12:47:06.907 SB.CCM firewallConnectCall: Creating association for traffic destined to 24.173.XX.XX:10048 for RTP

              12:47:06.907 SB.CCM firewallConnectCall: The association does not need NAT

              12:47:06.907 SB.CCM Delete criteria: Src: 216.16.xx.xx Dst: 24.173.XX.XX:10048 Vrf: 0 Proto: 17 Dir: Public

              12:47:06.908 SB.CCM firewallConnectCall: Creating association for traffic destined to 24.173.XX.XX:10049 for RTCP

              12:47:06.908 SB.CCM firewallConnectCall: The association does not need NAT

              12:47:06.909 SB.CCM Delete criteria: Src: 216.16.xx.xx Dst: 24.173.XX.XX:10049 Vrf: 0 Proto: 17 Dir: Public

              12:47:06.909 SB.CCM firewallConnectCall: Creating association for traffic destined to 216.16.XX.XX:38672 for RTP

              12:47:06.909 SB.CCM firewallConnectCall: The association does not need NAT

              12:47:06.909 SB.CCM Delete criteria: Src: 24.173.XX.XX Dst: 216.16.XX.XX:38672 Vrf: 0 Proto: 17 Dir: SELF

              12:47:06.909 SB.CCM firewallConnectCall: Creating association for traffic destined to 216.16.XX.XX:38673 for RTCP

              12:47:06.910 SB.CCM firewallConnectCall: The association does not need NAT

              12:47:06.910 SB.CCM Delete criteria: Src: 24.173.27.194:0 Dst: 216.16.XX.XX:38673 Vrf: 0 Proto: 17 Dir: SELF

              12:47:06.910 SB.CALL 15 State change      >> PreConnecting->PreConnected

              12:47:06.911 SB.CALL 15 PreConnected         Call PreConnecting from 115 to T06

              12:47:06 SB.CallStructObserver 15 <-> 558c0a8-18ad1bc2-13c4-483e3-807e73ca-483e3@bsip.bestline.netbcs adtran 7100 demo#

              12:47:12.081 SB.CALL 15 PreConnected         Called the clearCall routine

              12:47:12.081 SB.CALL 15 PreConnected         ClearCall sent from T06 to 115

              12:47:12.081 SB.CALL 15 State change      >> PreConnected->Clearing

              12:47:12.082 SB.CALL 15 Clearing             Called the clearResponse routine

              12:47:12.082 SB.CALL 15 State change      >> Clearing->CallIdlePending

              12:47:12.082 SB.CCM disconnect:

              12:47:12.082 SB.CCM  :  Call Struct 0x4134010 :   Call-ID = 15

              12:47:12.083 SB.CCM  :  Org Acct = 115    Dst Acct = T06

              12:47:12.083 SB.CCM  :  Org Port ID = 0/0.0   Dst Port ID = 0/0.65

              12:47:12.083 SB.CCM  :  Org TID =    Dst TID =

              12:47:12.083 SB.CCM  :  SDP Transaction = CallID: 15

              12:47:12.083 SB.CCM  :  SDP Offer = 0x03d8b710, (127.0.0.1:10048)

              12:47:12.083 SB.CCM  :  SDP Answer = 0x03d8ab10, (216.16.XX.XX:38672)

              12:47:12.084 SB.CCM disconnect: Call Connection Type is RTP_TO_RTP

              12:47:12.084 SB.CCM disconnect, music on hold

              12:47:12.084 SB.CCM firewallDisconnectCall: No action taken, call has already been cleaned up

              12:47:12.084 SB.CCM release:

              12:47:12.084 SB.CCM  :  Call Struct 0x4134010 :   Call-ID = 15

              12:47:12.084 SB.CCM  :  Org Acct = 115    Dst Acct = T06

              12:47:12.084 SB.CCM  :  Org Port ID = 0/0.0   Dst Port ID = 0/0.65

              12:47:12.085 SB.CCM  :  Org TID =    Dst TID =

              12:47:12.085 SB.CCM  :  SDP Transaction = CallID: 15

              12:47:12.085 SB.CCM  :  SDP Offer = 0x03d8b710, (127.0.0.1:10048)

              12:47:12.085 SB.CCM  :  SDP Answer = 0x03d8ab10, (216.16.XX.XX:38672)

              12:47:12.085 SB.CCM release: Call Connection Type is RTP_TO_RTP

              12:47:12.086 SB.CALL 15 CallIdlePending      ClearResponse sent from 115 to T06


              Any ideas?  The phone I'm calling does not ring though.

                • Re: Voice Loopback
                  Employee

                  I need to see the output from a debug sip stack messages and debug voice verbose, both enabled at the same time when a test call fails.  Can you submit those along with the current configuration to our FTP server with the instructions below?

                   

                  Open Internet Explorer web browser on their PC
                  Type the following URL:  ftp://ftp.adtran.com

                  Press the Alt key, click View, and then click Open FTP Site in Windows Explorer

                  Double-click the "Incoming" folder
                  Drag and drop files from PC into the Internet Explorer window

                  Reply to this post with the exact filenames used so we can retrieve the files


                  Thanks,

                  Matt

                    • Re: Voice Loopback
                      touristsis Visitor


                      Thanks Matt.

                      There are two files on incoming.  config-bcs-adtran....cfg and loopback.txt

                        • Re: Voice Loopback
                          Employee

                          Thanks for providing those.  The debug showed the SIP provider responded with a SIP/2.0 484 Address Incomplete.  Try initiating the call without a preceding 9, get a new debug to accompany that call attempt, and submit them to the FTP server.

                           

                          Thanks,

                          Matt

                            • Re: Voice Loopback
                              touristsis Visitor

                              Initiating loopback call id 23.

                              Hi Matt, without the 9, here is what I'm getting.

                               

                              2013.09.13 12:56:27 VOICELOOPBACK.115 Loopback account 115 failed CLI call to invalid number 512695XXXX

                                • Re: Voice Loopback
                                  Employee

                                  The SIP provider will need to give an explanation on why they responded with a "SIP/2.0 484 Address Incomplete" in the previous call attempt.  I would suggest supplying the debug you uploaded to them to see if they can resolve the issue or offer any insight.  It would also be a useful data point to try and dial the same number from a regular phone on site instead of the loopback account to make sure that works.  If you can get a debug of that I would be happy to review it, but the SIP provider should still be contacted to explain the "SIP/2.0 484 Address Incomplete".

                                   

                                  Thanks,

                                  Matt

                      • Re: Voice Loopback
                        Employee

                        I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily. If you don't feel like the answer I marked was correct, feel free to come back to this post to unmark it and select another in its place with the applicable buttons.  If you have any additional information on this that others may benefit from, please come back to this post to provide an update.  If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.

                         

                        Thanks,

                        Matt