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Unread 09-16-2007, 03:56 PM   #1
richard
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Default Does anyone have Incoming Calls working with a Voxbone, LES.NET or CallCentric DID?

Hi

I've tested with 2 SPA1001s and 1 PAP2. One of the SPA1001 are even on a different home network. All are located in the DMZ.

The result of any calls coming in via these DIDs is ONE-WAY audio. All other DID provider numbers work fine.

My ATAs are all registered to us.voxalot.com.

I have tried making a few changes to NAT settings, but no luck.

If anyone has DIDs working from these number. Either your own dedicated DID or through a SIPBroker gateway DID for these DID providers, would you kindly share your ATA configuration details? (or solution)

Thanks in advance


Richard
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Unread 09-17-2007, 02:32 PM   #2
baikal
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Hi

I had no problems with les.net, using either SIPBroker gateway DIDs and *010xxxxxx or my own DID at les.net. My DID is sip-forwarded to xxxxxx@voxalot.com in the les.net trunk setup, the ATA is a PAP2T registered to us.voxalot.com. I will have a look at its configuration when I come home ...

Marcel
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Unread 09-18-2007, 02:37 AM   #3
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Thanks Baikal. Hopefully your configuration will help sort this out. Richard
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Unread 09-19-2007, 03:45 AM   #4
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Below my settings (SIP & Line1 tab of the PAP2t configuration page, les.net trunk setup). Which way is your one-way audio? I recently observed one-way audio (no outgoing audio!) with another provider which is registered in my voxalot account.

PAP2T-SIP:
Code:
       SIP    
Parameters
               Max        [70             ] Max          [5              ]
               Forward:                     Redirection:
               Max Auth:  [2              ] SIP User     [$VERSION       ]
                                            Agent Name:
               SIP Server [$VERSION       ] SIP Reg User [               ]
               Name:                        Agent Name:
               SIP Accept [               ] DTMF Relay   [application/dtmf-relay]
               Language:                    MIME Type:
               Hook Flash [application/hook-flash] Remove Last  [no ]
               MIME Type:                   Reg:
               Use                          Escape
               Compact    [no ]             Display      [no ]
               Header:                      Name:
 SIP Timer
    Values    
     (sec)
               SIP T1:    [.5             ] SIP T2:      [4              ]
               SIP T4:    [5              ] SIP Timer B: [32             ]
               SIP Timer  [32             ] SIP Timer H: [32             ]
               F:
               SIP Timer  [32             ] SIP Timer J: [32             ]
               D:
               INVITE     [240            ] ReINVITE     [30             ]
               Expires:                     Expires:
               Reg Min    [1              ] Reg Max      [7200           ]
               Expires:                     Expires:
               Reg Retry  [30             ] Reg Retry    [1200           ]
               Intvl:                       Long Intvl:
  Response
    Status    
      Code
  Handling
               SIT1 RSC:  [               ] SIT2 RSC:    [               ]
               SIT3 RSC:  [               ] SIT4 RSC:    [               ]
               Try Backup [               ] Retry Reg    [               ]
               RSC:                         RSC:
       RTP    
Parameters
               RTP Port   [16384          ] RTP Port     [16482          ]
               Min:                         Max:
               RTP Packet [0.030          ] Max RTP ICMP [0              ]
               Size:                        Err:
               RTCP Tx    [0              ] No UDP       [no ]
               Interval:                    Checksum:                         
               Stats In   [no ]
               BYE:
       SDP
   Payload    
     Types
               NSE                          AVT Dynamic
               Dynamic    [100            ] Payload:     [101            ]
               Payload:
               INFOREQ                      G726r16
               Dynamic    [               ] Dynamic      [98             ]
               Payload:                     Payload:
               G726r24                      G726r40
               Dynamic    [97             ] Dynamic      [96             ]
               Payload:                     Payload:
               G729b                        NSE Codec
               Dynamic    [99             ] Name:        [NSE            ]
               Payload:
               AVT Codec  [telephone-event] G711u Codec  [PCMU           ]
               Name:                        Name:
               G711a                        G726r16
               Codec      [PCMA           ] Codec Name:  [G726-16        ]
               Name:
               G726r24                      G726r32
               Codec      [G726-24        ] Codec Name:  [G726-32        ]
               Name:
               G726r40                      G729a Codec
               Codec      [G726-40        ] Name:        [G729a          ]
               Name:
               G729b                        G723 Codec
               Codec      [G729ab         ] Name:        [G723           ]
               Name:
       NAT
   Support    
Parameters
               Handle VIA [no ]             Handle VIA   [no ]
               received:                    rport:
               Insert VIA [no ]             Insert VIA   [no ]
               received:                    rport:
               Substitute [yes]             Send Resp To [no ]
               VIA Addr:                    Src Port:
               STUN       [yes]             STUN Test    [no ]
               Enable:                      Enable:
               STUN       [stun.voxalot.com.au] EXT IP:      [               ]
               Server:
               EXT RTP    [               ] NAT Keep     [15             ]
               Port Min:                    Alive Intvl:
PAP2T-Line1:
Code:
    Streaming
 Audio Server    
        (SAS)
                  SAS Enable:    [no ]             SAS DLG Refresh  [30             ]
                                                   Intvl:
                  SAS Inbound    [               ]
                  RTP Sink:
 NAT Settings    
                  NAT Mapping    [yes]             NAT Keep Alive   [yes]
                  Enable:                          Enable:
                  NAT Keep Alive [$NOTIFY        ] NAT Keep Alive   [$PROXY         ]
                  Msg:                             Dest:
      Network    
     Settings
                  SIP TOS/                         Network Jitter
                  DiffServ       [0x68           ] Level:           [high          ]
                  Value:
                  RTP TOS/                         Jitter Buffer
                  DiffServ       [0xb8           ] Adjustment:      [up and down]
                  Value:
 SIP Settings    
                  SIP Port:      [5060           ] SIP 100REL       [no ]
                                                   Enable:
                  EXT SIP Port:  [               ] Auth             [yes]
                                                   Resync-Reboot:
                  SIP            [               ] SIP              [no ]
                  Proxy-Require:                   Remote-Party-ID:
                  SIP GUID:      [no ]             SIP Debug        [1-line                   ]
                                                   Option:
                  RTP Log Intvl: [0              ] Restrict Source  [no ]
                                                   IP:
                  Referor Bye    [4              ] Refer Target Bye [0              ]
                  Delay:                           Delay:
                  Referee Bye    [0              ] Refer-To Target  [no ]
                  Delay:                           Contact:
                  Sticky 183:    [no ]
 Call Feature    
     Settings
                  Blind
                  Attn-Xfer      [no ]             MOH Server:      [               ]
                  Enable:
                  Xfer When      [yes]             Conference       [               ]
                  Hangup Conf:                     Bridge URL:
                  Conference     [3 ]
                  Bridge Ports:
    Proxy and    
 Registration
                  Proxy:         [us.voxalot.com ] Use Outbound     [no ]
                                                   Proxy:
                  Outbound       [               ] Use OB Proxy In  [no ]
                  Proxy:                           Dialog:
                  Register:      [yes]             Make Call        [no ]
                                                   Without Reg:
                  Register       [600            ] Ans Call Without [yes]
                  Expires:                         Reg:
                  Use DNS SRV:   [yes]             DNS SRV Auto     [yes]
                                                   Prefix:
                  Proxy Fallback [600            ] Proxy Redundancy [Based on SRV Port]
                  Intvl:                           Method:
                  Voice Mail     [us.voxalot.com ]
                  Server:
   Subscriber    
  Information
                  Display Name:  [ABCD           ] User ID:         [nnnnnn         ]
                  Password:      [*************  ] Use Auth ID:     [yes]
                  Auth ID:       [nnnnnn         ]
                  Mini           [                                                  ]              
                  Certificate:
                  SRTP Private   [                                                  ]
                  Key:
Supplementary
      Service    
 Subscription
                  Call Waiting   [yes]             Block CID Serv:  [yes]
                  Serv:
                  Block ANC      [yes]             Dist Ring Serv:  [yes]
                  Serv:
                  Cfwd All Serv: [yes]             Cfwd Busy Serv:  [yes]
                  Cfwd No Ans    [yes]             Cfwd Sel Serv:   [yes]
                  Serv:
                  Cfwd Last      [yes]             Block Last Serv: [yes]
                  Serv:
                  Accept Last    [yes]             DND Serv:        [yes]
                  Serv:
                  CID Serv:      [yes]             CWCID Serv:      [yes]
                  Call Return    [yes]             Call Back Serv:  [yes]
                  Serv:
                  Three Way Call [yes]             Three Way Conf   [yes]
                  Serv:                            Serv:
                  Attn Transfer  [yes]             Unattn Transfer  [yes]
                  Serv:                            Serv:
                  MWI Serv:      [yes]             VMWI Serv:       [yes]
                  Speed Dial     [yes]             Secure Call      [yes]
                  Serv:                            Serv:
                  Referral Serv: [yes]             Feature Dial     [yes]
                                                   Serv:
                  Service
                  Announcement   [no ]
                  Serv:
        Audio    
Configuration
                  Preferred      [G729a  ]         Silence Supp     [no ]
                  Codec:                           Enable:
                  Use Pref Codec [no ]             Silence          [medium]
                  Only:                            Threshold:
                  G729a Enable:  [yes]             Echo Canc        [yes]
                                                   Enable:
                  G723 Enable:   [yes]             Echo Canc Adapt  [yes]
                                                   Enable:
                  G726-16        [yes]             Echo Supp        [yes]
                  Enable:                          Enable:
                  G726-24        [yes]             FAX CED Detect   [yes]
                  Enable:                          Enable:
                  G726-32        [yes]             FAX CNG Detect   [yes]
                  Enable:                          Enable:
                  G726-40        [yes]             FAX Passthru     [G711u]
                  Enable:                          Codec:
                  FAX Codec      [yes]             FAX Passthru     [NSE     ]
                  Symmetric:                       Method:
                  DTMF Tx        [Auto  ]          FAX Process NSE: [yes]
                  Method:
                  Hook Flash Tx  [None]            FAX Disable      [no ]
                  Method:                          ECAN:
                  Release Unused [yes]
                  Codec:
    Dial Plan    
                  Dial Plan:     [(x.|x*x.|**x.|*x.|<#:*>xxx.)                      ]
                  Enable IP      [no ]             Emergency        [               ]
                  Dialing:                         Number:
     FXS Port
     Polarity    
Configuration
                  Idle Polarity: [Forward]         Caller Conn      [Forward]
                                                   Polarity:
                  Callee Conn    [Forward]
                  Polarity:                                                                                           
               
                  Line Enable:   [yes]
trunk setup @ les.net:
Code:
                       Peer Name NNNNNNNNNNNNNNN   
                Your Description [forward to voxalot]   
              [?]Peer Technology [SIP ]   
                    [?]DTMF Mode [rfc2833]   
                    [?]Re-Invite [no ]   
                          [?]NAT [no ]   
                 [?]Error Method [Verbal]   
                       [?]Codecs[*]G.711   [*]G.729   [*]GSM   [*]G.726   
                    [?]Peer Type [URI         ] (URI/PSTN Experimental)   
                 [?]Peer Address [nnnnnn@voxalot.com  ]   
                     [?]Password [                    ]   
                   [?]Registered No   
                [?]Registered IP nnnnnn@voxalot.com   
         [?]Registration Expires No   
            [?]Outbound CallerID [                    ]   
              [?]7-Digit Dialing [ ]   
     [?]7-Digit Area Code Prefix [   ]   
   [?]10-Digit Dialing, Prefix 1 [ ]
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Unread 09-20-2007, 12:06 AM   #5
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My Voxbone DID (which is mapped to my Voxalot SIP URI) works fine, two-way audio.

NAT
Support
Parameters
Handle VIA [no ] Handle VIA [no ]
received: rport:
Insert VIA [no ] Insert VIA [no ]
received: rport:
Substitute [yes] Send Resp To [no ]
VIA Addr: Src Port:
STUN [yes] STUN Test [no ]
Enable: Enable:
STUN [stun.voxalot.com.au] EXT IP: [ ]
Server:
EXT RTP [ ] NAT Keep [15 ]
Port Min: Alive Intvl:

I recommend changing "Send Resp To" to Yes. Normally it should be Yes if "Substitute VIA Addr" is Yes. That might help.

If not try changing the other four VIA settings to Yes and those two back to No (either the top four should be yes, or the fifth and sixth ones should be yes, not both at once--I am told).

If still no luck, DMZ your PAP.
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Unread 09-20-2007, 04:22 AM   #6
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Thanks Baikal and Ctaylor.

I'll try playing with the setting again tomorrow and let you know.

BTW, my device is in the DMZ already.

Take care

Richard
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Unread 09-20-2007, 02:00 PM   #7
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I have the weird relationship of putting my ATA in the DMZ of my router, while keeping the various NAT compensating/port-busting techniques of the ATA enabled (in my case the fifth and six options to Yes), actually kills my device and cuts out voices. Just tried again now, and still have that happen, so I have to leave off the DMZ mode unless I want to investigate further what the settings should be.

I recommend playing with the STUN settings while not having any dmz or special port forwarding rules enabled for your ATA.
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