Smartnode justvoip caller id not displayed

botta · 2 Apr 2008

Hello,
i am using a smartnode 1200 (isdn) for outgoing calls on voip with justvoip provider.

When i make a call, the receiver don't see my phone number (privé/privat/unknown).

Under justvoip (betamax group) , i have setted and verified the "caller id" (im my case +41217778899 (not possible to set 0041217778899)).

Please help,

Remi

My smartnode sip config :

Code:

gateway sip GW_SIP_XXX
  bind interface IF_IP_WAN router
  no shutdown
  registrar sip.justvoip.com
  user +41217778899
  registration-lifetime 300
  domain sip.justvoip.com
  default-server sip.justvoip.com
  authentication sip.justvoip.com MyLogin MyPassword default

The log when calling a phone number (0778765432) :

Code:

192.168.1.241#20:58:01  SIP_SI> [EP IF_SIP_XXX-007fa398 SES 0x7fb880] > Stack: INVITE sip:[email protected]
20:58:01  SIP_SI> [EP IF_SIP_XXX-007fa398 SES 0x7fb880]          From:                    sip:[email protected]
20:58:01  SIP_SI> [EP IF_SIP_XXX-007fa398 SES 0x7fb880]          To:                      sip:[email protected]
20:58:01  SIP_TR> [GW] > Stack: INVITE sip:[email protected] SIP/2.0
20:58:01  SIP_TR> [GW] < Stack: SIP/2.0 401 Unauthorized
20:58:01  SIP_TR> [GW] > Stack: ACK sip:[email protected] SIP/2.0
20:58:01  SIP_SI> [EP IF_SIP_XXX-007fa398 SES 0x7fb880] < Stack: Authentication Reqired: sip.justvoip.com
20:58:01  SIP_SI> [GW GW_SIP_XXX] Using authentication data from user:'MyLogin' on service:'default'
20:58:01  SIP_TR> [GW] > Stack: INVITE sip:[email protected] SIP/2.0
20:58:01  SIP_TR> [GW] < Stack: SIP/2.0 100 Trying
20:58:01  SIP_SI> [EP IF_SIP_XXX-007fa398 SES 0x7fb880] < Stack: 100 Trying
20:58:01  SIP_TR> [GW] < Stack: SIP/2.0 183 Session progress
20:58:01  SIP_SI> [EP IF_SIP_XXX-007fa398 SES 0x7fb880] < Stack: 183 Session progress
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 DP] Using datapath termination 0x0200000f
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 DP] Using VoIP profile:
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO] Configuring datapath termination: Voice
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Highpass Filter: enabled
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Post Filter: enabled
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Silence Supression: disabled
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Voice Update Frames: disabled
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   DTMF Relay: enabled
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Mute Encoder: enabled
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Modem Transmission mode: None
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax Transmission mode: None
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax Bypass Coder: undefined
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Modem Bypass Coder: undefined
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax Dejitter Max. Delay: 200
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Modem Dejitter Max. Delay: 200
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax Max. Bit Rate: 14400
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax Error Correction Mode: enabled
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax/Data HDLC: enabled
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax Detection Mode: CED-Tone
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax Bypass Method: Default
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Modem Bypass Method: Default
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Fax Volume: -9500
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO] Configuring datapath termination: Dejitter
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Dejitter Mode: adaptive
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Max. Delay: 60
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Max. Packet Loss: 4
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Shrink Speed: 1
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Grow Step: 1
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Grow Attenuation: 1
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Max. Delay: 60
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 DP] Peer call-leg changed to state CONNECTED
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 DP] Local Media Address: 192.168.1.241
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 DP] Add termination RTP-00/000f to context 0087f188
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO] Configuring datapath termination: RTP
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Local Address: 192.168.1.241/4864
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Remote Address: 194.120.0.191/41068
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Codec: G.711 u-law (20 ms)
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Media Type: audio
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   Payload: Voice=0, SID=13, NTE-Local=101, NTE-Remote=101, NSE-Local=255, NSE-Remote=255
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   SSRC: 11148056
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO]   TOS: 0
20:58:01  SIP_DP> [EP IF_SIP_XXX-007fa398/0 AUDIO] Datapath: Change direction to send/receive.
20:58:07  SIP_DP> [EP IF_SIP_XXX-007fa398/0 DP] Peer call-leg changed to state DISCONNECTING
20:58:07  SIP_DP> [EP IF_SIP_XXX-007fa398/0 DP] Subtract termination RTP-00/000f from context 0087f188
20:58:07  SIP_SI> [EP IF_SIP_XXX-007fa398 SES 0x7fb880] > Stack: BYE
20:58:07  SIP_TR> [GW] > Stack: CANCEL sip:[email protected] SIP/2.0
20:58:07  SIP_TR> [GW] < Stack: SIP/2.0 200 Ok
20:58:07  SIP_TR> [GW] < Stack: SIP/2.0 487 Request terminated
20:58:07  SIP_ER> [EP IF_SIP_XXX-007fa398 SES 0x7fb880] < Stack: UNHANDLED STACK EVENT: BASIC STATUS: 200 Ok
20:58:07  SIP_TR> [GW] > Stack: ACK sip:[email protected] SIP/2.0
20:58:07  SIP_SI> [EP IF_SIP_XXX-007fa398 SES 0x7fb880] < Stack: 487 Request terminated
20:58:07  SIP_SI> [EP IF_SIP_XXX-007fa398] Finished

The sip status :

Code:

SIP Gateway: GW_SIP_XXX
=======================

 State:                                 Up
 User Agent:                            0xa7f8a0
 Local Address:                         192.168.1.241:5060
 Registrations

 SIP Registration: sip:[email protected] (Service: default)
 ----------------------------------------------------------------------

  State:                                Registered
  Registrar:                            sip:sip.justvoip.com
  Default Server:                       (none)
  Logical Address:                      sip:[email protected]
  Physical Address:                     sip:[email protected]:5060 (Current: sip:[email protected]:5060)
  Expiration Time:                      300 s
  Authenticated:                        yes

 Allocated Endpoints:                   0
 Allocated Sessions:                    0
 Allocated Audio Datapath Controllers:  0
 Allocated Packets:                     1
 Max Allocated Packets:                 11
 Max Allocated Timers:                  9

Edit Guard-X: Please use Code-Tags!

botta · 7 Apr 2008

Hello,
i am using a smartnode 1200 (isdn) for outgoing calls on voip with justvoip provider.

When i make a call, the receiver don't see my phone number (privé/privat/unknown).

Under justvoip (betamax group) , i have setted and verified the "caller id" (im my case +41217778899 (not possible to set 0041217778899)).

With justvoip windows program = soft-phone, the receiver see my phone number setted and verified under justvoip.

It's a smartnode configuration problem ...

Any idea ?

Remi

botta · 13 Apr 2008

Hello,

Nobody uses the smartnode with justvoip or other Betamax SIP provider ?

Remi

botta · 3 Mai 2008

Hallo,

i have found a solution for smartnode :

justvoip account : MyLogin
justvoip password : MyPassword
my verified number on justvoip : 0041217778899
(with software justvoip the verified number is +41217778899)

I think is the same solution for all betamax voip providers.

Code:

.....
  interface sip IF_SIP_XXX
    bind gateway GW_SIP_XXX
    service default
    route call dest-interface IF_S0_PHONE
    address-translation outgoing-call from-header user-part fix 0041217778899 host-part call
    remote-party-id called-party 
    remote-party-id calling-party
    use profile voip PROF_SIP

gateway sip GW_SIP_XXX
  bind interface IF_IP_WAN router
  no shutdown
  registrar sip.justvoip.com
  user 0041217778899
  domain sip.justvoip.com
  authentication sip.justvoip.com MyLogin MyPassword default
  remote-party-id

......

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