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GXW4104 - detectie busy tone

synologic — GMT

Am un echipament, modelul din subiect si am ceva probleme cu detectia de busy tone, in consecinta nu se face hangup automat la linie.

Momentan testez cu o linie de RDS si am facut urmatoarele setari:

1. Enable Current Disconnect(Y/N): ch1-4:N;
2. Enable Tone Disconnect(Y/N): ch1-4:Y;
7. AC Termination Impedance: ch1-4:1;

1. Wait for Dial-Tone(Y/N): ch1-4:N;
2. Stage Method(1/2): ch1-4:1;

Busy Tone: ch1-4:f1=440@-17,f2=450@-17,c=180/160;

2. Rx from PSTN Audio Gain(dB): ch1-4:0; (am incercat si cu 6 si cu 9 si cu 12)

Bun, pana aici am enumerat ce am tot testat/modificat, fara nici un rezultat.
Apelurile functioneaza ok atat inbound cat si outbound (pbx asterisk gw-ul setat in mod peer)

Cum am obtinut valorile de la Busy Tone ... in doua moduri. Prima a fost variata manuala, am inregistrat un apel cu busy tone si m-am uitat cu un analizor de spectru, care imi dadea peak 450Hz, -17.5dB cu interval on/off 170ms
A doua varianta, am obtinut-o cu logging CPT Cadence Syslog din CLI-ul echipamentului, ce produce un output cam asa:

May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 160 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:35 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:35 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:35 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:35 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:35 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:35 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:36 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:36 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:36 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:36 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.

De unde si valorile, F1-440Hz, F2-450Hz, 176/160, am incercat si cu 180/160 sa fie "rotund", tot degeaba.

Ori nu functioneaza tone detection-ul, ori nu inteleg ce se intampla, in schimb, dupa ce activez debug pe syslogging, am observat o chestie dubioasa:

---- syslog imediat dupa reboot -----
May 12 20:32:51 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] adjust_dns_servers(): s_count=2; s0=8.8.8.8; s1=8.8.4.4
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Grandstream GXW4104 1.3.4.13 1.1.3.2
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] DNS response received.
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Record found and about to process.
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Start processing DNS response message.
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] SUCCESSFUL. result_ip is not 0; call callback function.
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Processing (Transitioning State), From: 777 , GWSt: 0
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 0 APP: app_State:last_fxo_call_status (0:0)
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Set State to: On-Hook, From: 851
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [0] from Line:700
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Dispatching (Idle), From:717
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET: Port: 0 Reset Tone Detection
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_DTMF: Port: 0 Disable DTMF Detection.
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Disable CPT Detection, tone mask: 000F
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO_CID: Port: 0 Disabling detection, Type: BELLCORE
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 1 Processing (Transitioning State), From: 777 , GWSt: 0
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 1 APP: app_State:last_fxo_call_status (0:0)
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 1 Set State to: On-Hook, From: 851
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 1 Stop tone [0] from Line:700
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 1 *Call Status* Dispatching (Idle), From:717
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET: Port: 1 Reset Tone Detection
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_DTMF: Port: 1 Disable DTMF Detection.
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 1 Disable CPT Detection, tone mask: 000F
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO_CID: Port: 1 Disabling detection, Type: BELLCORE
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 2 Processing (Transitioning State), From: 777 , GWSt: 0
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 2 APP: app_State:last_fxo_call_status (0:0)
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 2 Set State to: On-Hook, From: 851
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 2 Stop tone [0] from Line:700
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 2 *Call Status* Dispatching (Idle), From:717
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET: Port: 2 Reset Tone Detection
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_DTMF: Port: 2 Disable DTMF Detection.
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 2 Disable CPT Detection, tone mask: 000F
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO_CID: Port: 2 Disabling detection, Type: BELLCORE
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 3 Processing (Transitioning State), From: 777 , GWSt: 0
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 3 APP: app_State:last_fxo_call_status (0:0)
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 3 Set State to: On-Hook, From: 851
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 3 Stop tone [0] from Line:700
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 3 *Call Status* Dispatching (Idle), From:717
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET: Port: 3 Reset Tone Detection
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_DTMF: Port: 3 Disable DTMF Detection.
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 3 Disable CPT Detection, tone mask: 000F
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO_CID: Port: 3 Disabling detection, Type: BELLCORE
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Processing (Idle), From:1745
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 1 *Call Status* Processing (Idle), From:1745
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 2 *Call Status* Processing (Idle), From:1745
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 3 *Call Status* Processing (Idle), From:1745
May 12 20:32:52 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Line Event triggered is: 1 Previous Event was: 3 current line Status is: 4 count: 8
May 12 20:32:54 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Provision attempt 1
May 12 20:32:54 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] DNS response received.
May 12 20:32:54 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Record found and about to process.
May 12 20:32:54 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Start processing DNS response message.
May 12 20:32:54 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] SUCCESSFUL. result_ptr and callback are not NULL; call callback function.
May 12 20:32:54 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FW_UPGRADE: Lock download buffer.
May 12 20:33:02 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FW_UPGRADE: Unlock download buffer.

Nimic dubios pana aici, am configurat toate 4 liniile dar doar una are cablul conectat.

urmeaza niste linii cu SIP Registration, care sunt ok, dupa care, fac un apel (de pe clientul sip spre FXO):

May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Initiate Pick-Up, From: 1702
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] LEC_FXO: Port: 0 Enable.
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 402 sip.c Sess: 0 allocated for Port(acct): 0 ln#: 961 Grandstream GXW4104 1.3.4.13 1.1.3.2
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1269 sip_transport.c Sess: 0 Sending 100 To 192.168.10.30:5060 0 1 0 deflt
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1602 sip_transport.c Sess: 0 sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP ......
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1011 sip_handle_invite.c Sess: 0 INVITE .....
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] ------------------------- GXW setup ------------------------------
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1107 sip_handle_invite.c Port: 0 Sess: 0 Acc:0; FullRingCounts:0 RingPulseCounts:0 FXOState:3; Log user data: 1 StageDialing, PR Disabled; Curr.Disc. Disabled; Cont'd..
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] BZ TD Enabled,Tone Cfg: Dial: F: 440 440,V: 30 30,Cad: 0 0 0,0 0 0; Ringback: F: 440 425,V: 11 11,Cad: 1000 0 0,4000 0 0; BZ: F: 440 450,V: 17 17,Cad: 180 0 0,160 0 0; Reorder: F: 450 450,V: 30 30,Cad: 8 0 0,8 0 0; Cont'd..
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Caller ID Type: 1, Relay Type: 1, Wait for Dial tone: 0, Dialing Cfg: ch1-4:N;, Delay2Dial: 500; FAX TD Disable, T38: Enabled, T38 Bitrate:9600, T38 Mode:1; T38 ECM:1; Echo Can.: Y, Sil. Supp.: Y; Cont'd..
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Audio Gains: RX from PSTN: 0, TX to PSTN: 1, DTMF Method: 1, SIP port: 5060, SRTP: 0,; DTMF Interkey T.O: 4, Round-robin: 1, Cont'd..
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] Prefix Code: Len:2, Scheduling:99 Ring Count:4 RingTimeout:6 FAS:0 HW DS:0 Grandstream GXW4104 1.3.4.13 1.1.3.2 ProxyCallOnly:101
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] ------------------------- GXW setup (end)-------------------------
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] dialplan_process: segments checked = 1, result = DIALPLAN_MATCH_SUCCESS, match_index = 0
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1239 sip_handle_invite.c Port: 0 Sess: 0 Wait Dialtone: 0 Dial: When FXO is ready
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Processing (Off-Hook), From: 777 , GWSt: 3
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 0 APP: app_State:last_fxo_call_status (3:0)
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Set State to: Settling, From: 352
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Dispatching (Picking-up), From:880
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Processing (Picking-up), From:1745
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] cm_task.c 1773 Port: 0 GWSt: 3 Sess: 0 3 fxoCallStatus last/now=0/2 FAS:0
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Line Event triggered is: 3 Previous Event was: 1 current line Status is: 3 count: 8
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Processing (Transitioning State), From: 777 , GWSt: 3
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 0 APP: app_State:last_fxo_call_status (3:2)
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Set State to: Off-Hook, From: 962
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_DTMF: Port: 0 Enable DTMF detection.
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Enable CPT detection, tone mask: 0004
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Enable CPT detection, tone mask: 0008
May 12 20:24:14 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_VAD: Port: 0 Enable VAD discriminator.
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Processing (Transitioning State), From: 777 , GWSt: 3
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 0 APP: app_State:last_fxo_call_status (3:2)
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Dispatching (Ready for Dialing), From:1285
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Disable CPT Detection, tone mask: 0001
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Processing (Ready for Dialing), From:1745
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] cm_task.c 1773 Port: 0 GWSt: 3 Sess: 0 3 fxoCallStatus last/now=2/6 FAS:0
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1940 cm_utility.c Port: 0 Sess: 0 FXO_Call: 0(l:100 p:100 v:11)3(l:100 p:100 v:11)3(l:100 p:100 v:11)9(l:100 p:100 v:11)8(l:100 p:100 v:11)0(l:100 p:100 v:11)0(l:100 p:100 v:11)9(l:100 p:100 v:11)0(l:100 p:100 v:11)0(l:100 p:100 v:11)
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [51] from Line:672
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [51] f1:941 f2:1336 ON:10 OFF:10 Vol:13 11
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [51] from Line:795
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [33] from Line:751
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [33] f1:697 f2:1477 ON:10 OFF:10 Vol:13 11
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Processing (Line Activity Detected), From: 777 , GWSt: 4
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 0 APP: app_State:last_fxo_call_status (4:6)
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 WARNING!!! Detecting Voice !!! Initial dialing still not finished.
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [33] from Line:795
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [33] from Line:751
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [33] f1:697 f2:1477 ON:10 OFF:10 Vol:13 11
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [33] from Line:795
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [44] from Line:751
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [44] f1:852 f2:1477 ON:10 OFF:10 Vol:13 11
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [44] from Line:795
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [43] from Line:751
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [43] f1:852 f2:1336 ON:10 OFF:10 Vol:13 11
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [43] from Line:795
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [51] from Line:751
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [51] f1:941 f2:1336 ON:10 OFF:10 Vol:13 11
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [51] from Line:795
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [51] from Line:751
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [51] f1:941 f2:1336 ON:10 OFF:10 Vol:13 11
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [51] from Line:795
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [44] from Line:751
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [44] f1:852 f2:1477 ON:10 OFF:10 Vol:13 11
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [44] from Line:795
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [51] from Line:751
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [51] f1:941 f2:1336 ON:10 OFF:10 Vol:13 11
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [51] from Line:795
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Dialing:
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Start tone [51] from Line:751
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Details for tone [51] f1:941 f2:1336 ON:10 OFF:10 Vol:13 11
May 12 20:24:16 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [51] from Line:795
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Processing (Dialing Finished), From: 777 , GWSt: 4
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 0 APP: app_State:last_fxo_call_status (4:6)
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Dispatching (Dial OK), From:1097
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_VAD: Port: 0 Suspend VAD discriminator.
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Dispatching (Remote Ringback), From:1119
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Processing (Dial OK), From:1745
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] cm_task.c 1773 Port: 0 GWSt: 4 Sess: 0 4 fxoCallStatus last/now=6/7 FAS:0
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Processing (Remote Ringback), From:1745
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] cm_task.c 1773 Port: 0 GWSt: 4 Sess: 0 4 fxoCallStatus last/now=7/9 FAS:0
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1269 sip_transport.c Sess: 0 Sending 180 To 192.168.10.30:5060 4 1 0 0 sip.Call-ID:0ee0f2301d51dc6f1c322a442083c524@192.168.10.30:5060
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1602 sip_transport.c Sess: 0 sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP .......
May 12 20:24:17 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] cm_task.c 1983 0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 Processing (Line Activity Detected), From: 777 , GWSt: 5
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] fxo_ctl.c Port: 0 APP: app_State:last_fxo_call_status (5:9)
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Dispatching (Remotely Answered), From:1202
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_VAD: Port: 0 Enable VAD discriminator.
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Processing (Remotely Answered), From:1745
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] cm_task.c 1773 Port: 0 GWSt: 5 Sess: 0 5 fxoCallStatus last/now=9/10 FAS:0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 2004 cm_task.c Port: 0 Sess: 0 Polarity Reversal:0 101 0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 266 sdp.c Sess: 0 m1-rtx:0@17334-RTP payload type: 0; acc:0 sipSt:4 ccSt:1 rtpStarted:0 rtpRTX:0 Port: 0 0 0 2 1 1 0 0 0 0 0 0 0;
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 631 sdp.c Port: 0 Sess: 0 m-updated: 192.168.10.30:17334 0 20 101 2 0 5004 0 0 re-open voc i= 0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1269 sip_transport.c Sess: 0 Sending 200 To 192.168.10.30:5060 4 1 0 0 sip.Call-ID:0ee0f2301d51dc6f1c322a442083c524@192.168.10.30:5060
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1602 sip_transport.c Sess: 0 sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK30a8788d;rport From: ....
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 266 sdp.c Sess: 0 m1-rtx:0@17334-RTP payload type: 0; acc:0 sipSt:5 ccSt:4 rtpStarted:0 rtpRTX:0 Port: 0 0 0 2 1 1 0 0 0 0 0 0 0;
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 631 sdp.c Port: 0 Sess: 0 m-updated: 192.168.10.30:17334 0 20 101 2 0 5004 0 0 re-open voc i= 0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 266 sdp.c Sess: 0 m1-rtx:0@17334-RTP payload type: 0; acc:0 sipSt:5 ccSt:4 rtpStarted:0 rtpRTX:0 Port: 0 0 0 2 1 1 0 0 0 0 0 0 0;
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 631 sdp.c Port: 0 Sess: 0 m-updated: 192.168.10.30:17334 0 20 101 2 0 5004 0 0 re-open voc i= 0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1269 sip_transport.c Sess: 0 Sending 200 To 192.168.10.30:5060 5 1 0 0 sip.Call-ID:0ee0f2301d51dc6f1c322a442083c524@192.168.10.30:5060
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1602 sip_transport.c Sess: 0 sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK30a8788d;rport From: ....
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_DTMF: Port: 0 Reporting DTMF digit 3
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] cm_task.c 2044 Port: 0 get digit 3,6 5
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1263 sip.c Acc:0 Received SIP message: ACK sip:1000@192.168.10.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK396e15bd;rport Max-Forwards: 70 From: .....
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 207 sip_dialog.c Sess: 0 st: 5 loclOg: 2 msg: 3 ccSt: 4 ccRB'd: 0 oprt: 0 sip.Call-ID MATCHED, len&id:51 0ee0f2301d51dc6f1c322a442083c524@192.168.10.30:5060
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 137 sip_handle_ack.c Port: 0 Sess: 0 0 5 0 0 0 4 0 192.168.10.30:17334
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 343 sip_handle_ack.c handle ACK, Sess(: 0, st:6, est'd 1); startM:1; ccSt:4; sndrcv:0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 752 cm_rtp.c Open VOC@ Port: 0 Sess: 0 Payload:0 Rtp 192.168.10.30:17334 0 20 101 2 5004 0 0 0 1 0 0 callSt: 5 0 0 0 6
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] VOC: Open Port: 0 coder: 0 para: 0 algorithm ch: 0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] cm_api.c 334 Port: 0 Sess: 0 Start fax_detector in 2.5s
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 1263 sip.c Acc:0 Received SIP message: ACK sip:1000@192.168.10.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK021e6c27;rport Max-Forwards: 70 From: ....
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 207 sip_dialog.c Sess: 0 st: 6 loclOg: 2 msg: 3 ccSt: 5 ccRB'd: 0 oprt: 0 sip.Call-ID MATCHED, len&id:51 0ee0f2301d51dc6f1c322a442083c524@192.168.10.30:5060
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 137 sip_handle_ack.c Port: 0 Sess: 0 1 6 0 0 0 5 0 192.168.10.30:17334
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] 343 sip_handle_ack.c handle ACK, Sess(: 0, st:6, est'd 1); startM:0; ccSt:5; sndrcv:0
May 12 20:24:20 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_GEN: Port: 0 Stop tone [0] from Line:1536

aici am activat CPT Cadence logging

May 12 20:24:24 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET: Port: 0 Syslog Debug Level: 4 CPT Cadence ON.
May 12 20:24:25 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 3984 ms.

aici am inchis telefonul (de pe linia fxo)

May 12 20:24:26 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 425Hz detected for 1008 ms.
May 12 20:24:33 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 7488 ms.
May 12 20:24:33 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 80 ms.
May 12 20:24:33 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 160 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:34 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.
May 12 20:24:35 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 freq : 440Hz and 450Hz detected for 176 ms.
May 12 20:24:35 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Silence detected: 160 ms.

Ce mi se pate foarte dubios, sunt liniile astea:

May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] FXO: Port: 0 *Call Status* Dispatching (Ready for Dialing), From:1285
May 12 20:24:15 192.168.10.32 GS_LOG: [00:0B:82:32:17:77][000][9660000320A][1.3.4.13] TONE_DET_CPT: Port: 0 Disable CPT Detection, tone mask: 0001

dupa care, nu mai scrie nicaieri nimic ca s-a enabled CPT Detection...

In consecinta, nu se detecteaza tonul de busy, ori am gresit eu ceva in config, ori in schema asta nu se seteaza asa, nu stiu ce sa zic, m-am tot chinuit astazi si nu mi-a iesit nimic.

Orice ajutor e binevenit :)

Problema NEC Univerge sv8100 - Asterisk SIP Trunk

skunky — GMT

Salut !
Am o MARE problema, ce trebuie rezolvata sau macar identificata.
Centrala NEC-Phillips cu licenta de trunchi SIP instalata si configurata, activat g711a.
Elastix - Asterisk 1.8.7 cu un trunchi sip creat, avand urmatoarele setari:

host=192.168.1.11
type=peer
disallow=all
allow=alaw

la incoming am lasat blank.

Are pus si "allow anonymous inbound calls".

(192.168.1.11 - NEC, 192.168.1.16 - Asterisk)

Apelez de pe un Aastra inregistrat in asterisk cu extensie 3801 catre 104 - extensie NEC. Suna, se aude, totul ok.

Apelez de pe 104 catre 3801 - 3801 suna - raspund - nu se aude nimic.
Apelez de pe 104 catre 3801 - 3801 suna - inchid 104 - 3801 continua sa sune la nesfarsit.

La un moment dat, in log apare "Ignoring this invite request - De ce oare, asta o fi cauza?"

Deja sunt in faza in care nu mai stiu ce sa-i fac, domnii de la NEC m-au ajutat pana la un punct in care mi-au precizat ca Asterisk nu figureaza pe lista de echipamente compatibile.
Totusi cred ca mie imi scapa ceva in cofigurarea Asterisk.

Pun si o bucata (cam mare) de log - apel 104 -> 3801 si raspuns pe 3801 (nu aud voce) - imi pare rau daca am postat prea multa informatie:

<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100";tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Sending to 192.168.1.11:5060 (no NAT)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Using INVITE request as basis request - 0201C1A90C81400000000010@192.168.1.11
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found peer 'nec-silf' for '100' from 192.168.1.11:5060
[May 8 13:17:27] VERBOSE[10572] netsock2.c: == Using SIP RTP TOS bits 184
[May 8 13:17:27] VERBOSE[10572] netsock2.c: == Using SIP RTP CoS mark 5
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 8
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 2
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 18
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 9
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format PCMA for ID 8
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G726-32 for ID 2
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G729 for ID 18
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G722 for ID 9
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Capabilities: us - 0x8 (alaw), peer - audio=0x1908 (alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Peer audio RTP is at port 192.168.1.20:10026
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Looking for 3801 in from-trunk-sip-nec-silf (domain 192.168.1.16)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: list_route: hop:
[May 8 13:17:27] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To:
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

<------------>
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [3801@from-trunk-sip-nec-silf:1] Set("SIP/nec-silf-00000004", "GROUP()=OUT_4") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [3801@from-trunk-sip-nec-silf:2] Goto("SIP/nec-silf-00000004", "from-trunk,3801,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (from-trunk,3801,1)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [3801@from-trunk:1] Macro("SIP/nec-silf-00000004", "exten-vm,novm,3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/nec-silf-00000004", "user-callerid,") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/nec-silf-00000004", "AMPUSER=100") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/nec-silf-00000004", "0?report") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/nec-silf-00000004", "1?Set(REALCALLERIDNUM=100)") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/nec-silf-00000004", "AMPUSER=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/nec-silf-00000004", "AMPUSERCIDNAME=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/nec-silf-00000004", "1?report") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-user-callerid,s,10)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/nec-silf-00000004", "0?continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:11] Set("SIP/nec-silf-00000004", "__TTL=64") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/nec-silf-00000004", "1?continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-user-callerid,s,19)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:19] Set("SIP/nec-silf-00000004", "CALLERID(number)=100") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:20] Set("SIP/nec-silf-00000004", "CALLERID(name)=100") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:21] NoOp("SIP/nec-silf-00000004", "Using CallerID "100" <100>") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/nec-silf-00000004", "RingGroupMethod=none") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/nec-silf-00000004", "VMBOX=novm") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/nec-silf-00000004", "__EXTTOCALL=3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/nec-silf-00000004", "CFUEXT=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/nec-silf-00000004", "CFBEXT=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/nec-silf-00000004", "RT=""") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/nec-silf-00000004", "record-enable,3801,IN") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/nec-silf-00000004", "1?check") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-record-enable,s,4)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/nec-silf-00000004", "0?MacroExit()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/nec-silf-00000004", "0?Group:OUT") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-record-enable,s,15)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/nec-silf-00000004", "1?IN") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-record-enable,s,20)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:20] ExecIf("SIP/nec-silf-00000004", "1?MacroExit()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/nec-silf-00000004", "dial-one,"",tr,3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:1] Set("SIP/nec-silf-00000004", "DEXTEN=3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:2] Set("SIP/nec-silf-00000004", "DIALSTATUS_CW=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:3] GosubIf("SIP/nec-silf-00000004", "0?screen,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:4] GosubIf("SIP/nec-silf-00000004", "0?cf,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:5] GotoIf("SIP/nec-silf-00000004", "1?skip1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,8)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:8] GotoIf("SIP/nec-silf-00000004", "0?nodial") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:9] GotoIf("SIP/nec-silf-00000004", "0?continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:10] Set("SIP/nec-silf-00000004", "EXTHASCW=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:11] GotoIf("SIP/nec-silf-00000004", "1?next1:cwinusebusy") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,12)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:12] GotoIf("SIP/nec-silf-00000004", "0?docfu:skip3") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,16)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:16] GotoIf("SIP/nec-silf-00000004", "1?next2:continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,17)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:17] GotoIf("SIP/nec-silf-00000004", "1?continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,25)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:25] GotoIf("SIP/nec-silf-00000004", "0?nodial") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:26] GosubIf("SIP/nec-silf-00000004", "1?dstring,1:dlocal,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:1] Set("SIP/nec-silf-00000004", "DSTRING=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:2] Set("SIP/nec-silf-00000004", "DEVICES=3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/nec-silf-00000004", "0?Return()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/nec-silf-00000004", "0?Set(DEVICES=801)") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:5] Set("SIP/nec-silf-00000004", "LOOPCNT=1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:6] Set("SIP/nec-silf-00000004", "ITER=1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:7] Set("SIP/nec-silf-00000004", "THISDIAL=SIP/3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/nec-silf-00000004", "1?zap2dahdi,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/nec-silf-00000004", "0?Return()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/nec-silf-00000004", "NEWDIAL=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/nec-silf-00000004", "LOOPCNT2=1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/nec-silf-00000004", "ITER2=1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/nec-silf-00000004", "THISPART2=SIP/3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/nec-silf-00000004", "0?Set(THISPART2=DAHDI/3801)") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/nec-silf-00000004", "NEWDIAL=SIP/3801&") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/nec-silf-00000004", "ITER2=2") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/nec-silf-00000004", "0?begin2") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/nec-silf-00000004", "THISDIAL=SIP/3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/nec-silf-00000004", "") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:9] Set("SIP/nec-silf-00000004", "DSTRING=SIP/3801&") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:10] Set("SIP/nec-silf-00000004", "ITER=2") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/nec-silf-00000004", "0?begin") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:12] Set("SIP/nec-silf-00000004", "DSTRING=SIP/3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:13] Return("SIP/nec-silf-00000004", "") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:27] GotoIf("SIP/nec-silf-00000004", "0?nodial") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:28] GotoIf("SIP/nec-silf-00000004", "1?skiptrace") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,30)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:30] Set("SIP/nec-silf-00000004", "D_OPTIONS=tr") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:31] ExecIf("SIP/nec-silf-00000004", "0?SIPAddHeader(Alert-Info: )") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:32] ExecIf("SIP/nec-silf-00000004", "0?SIPAddHeader()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:33] ExecIf("SIP/nec-silf-00000004", "0?Set(CHANNEL(musicclass)=)") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:34] GosubIf("SIP/nec-silf-00000004", "0?qwait,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:35] Set("SIP/nec-silf-00000004", "__CWIGNORE=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:36] Set("SIP/nec-silf-00000004", "__KEEPCID=TRUE") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:37] Dial("SIP/nec-silf-00000004", "SIP/3801,"",tr") in new stack
[May 8 13:17:27] VERBOSE[10685] netsock2.c: == Using SIP RTP TOS bits 184
[May 8 13:17:27] VERBOSE[10685] netsock2.c: == Using SIP RTP CoS mark 5
[May 8 13:17:27] VERBOSE[10685] app_dial.c: -- Called SIP/3801
[May 8 13:17:27] VERBOSE[10685] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

<------------>
[May 8 13:17:28] VERBOSE[10685] app_dial.c: -- SIP/3801-00000005 is ringing
[May 8 13:17:28] VERBOSE[10685] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

<------------>
[May 8 13:17:28] VERBOSE[10685] app_dial.c: -- SIP/3801-00000005 is ringing
[May 8 13:17:28] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100";tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:28] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:28] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:28] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To:
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

<------------>
[May 8 13:17:29] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100";tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:29] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:29] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:29] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To:
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

[May 8 13:17:30] VERBOSE[10685] app_dial.c: -- SIP/3801-00000005 answered SIP/nec-silf-00000004
[May 8 13:17:30] VERBOSE[10685] chan_sip.c: Audio is at 5060
[May 8 13:17:30] VERBOSE[10685] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 8 13:17:30] VERBOSE[10685] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:31] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100";tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:31] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To:
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0

<------------>
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Audio is at 5060
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 8 13:17:31] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763839 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
[May 8 13:17:32] VERBOSE[10572] chan_sip.c: Retransmitting #2 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:34] VERBOSE[10572] chan_sip.c: Retransmitting #3 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [h@macro-dial-one:1] Macro("SIP/nec-silf-00000004", "hangupcall,") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/nec-silf-00000004", "1?endmixmoncheck") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,9)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:9] NoOp("SIP/nec-silf-00000004", "End of MIXMON check") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/nec-silf-00000004", "1?nomeetmemon") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,15)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:15] NoOp("SIP/nec-silf-00000004", "MEETME_RECORDINGFILE=") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:16] GotoIf("SIP/nec-silf-00000004", "1?noautomon") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,18)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:18] NoOp("SIP/nec-silf-00000004", "TOUCH_MONITOR_OUTPUT=") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:19] GotoIf("SIP/nec-silf-00000004", "1?noautomon2") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,25)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:25] NoOp("SIP/nec-silf-00000004", "MONITOR_FILENAME=") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:26] GotoIf("SIP/nec-silf-00000004", "1?skiprg") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,29)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:29] GotoIf("SIP/nec-silf-00000004", "1?skipblkvm") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,32)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:32] GotoIf("SIP/nec-silf-00000004", "1?theend") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,34)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:34] Hangup("SIP/nec-silf-00000004", "") in new stack
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/nec-silf-00000004' in macro 'hangupcall'
[May 8 13:17:34] VERBOSE[10685] features.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/nec-silf-00000004'
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/nec-silf-00000004' in macro 'dial-one'
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/nec-silf-00000004' in macro 'exten-vm'
[May 8 13:17:34] VERBOSE[10685] pbx.c: == Spawn extension (from-trunk, 3801, 1) exited non-zero on 'SIP/nec-silf-00000004'
[May 8 13:17:34] VERBOSE[10685] chan_sip.c: Scheduling destruction of SIP dialog '0201C1A90C81400000000010@192.168.1.11' in 32000 ms (Method: INVITE)
[May 8 13:17:35] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100";tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:35] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:35] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:35] NOTICE[10572] chan_sip.c: Unable to create/find SIP channel for this INVITE
[May 8 13:17:35] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[May 8 13:17:35] VERBOSE[10572] chan_sip.c: Scheduling destruction of SIP dialog '0201C1A90C81400000000010@192.168.1.11' in 32000 ms (Method: INVITE)
[May 8 13:17:38] VERBOSE[10572] chan_sip.c: Retransmitting #4 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:42] VERBOSE[10572] chan_sip.c: Retransmitting #5 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:43] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100";tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:43] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:43] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:46] VERBOSE[10572] chan_sip.c: Retransmitting #6 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:50] VERBOSE[10572] chan_sip.c: Retransmitting #7 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:54] VERBOSE[10572] chan_sip.c: Retransmitting #8 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:58] VERBOSE[10572] chan_sip.c: Retransmitting #9 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

[May 8 13:17:59] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100";tag=338C324631353641000B6B8E
To:
Contact:
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:59] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:59] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request

[May 8 13:18:02] VERBOSE[10572] chan_sip.c: Retransmitting #10 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100";tag=338C324631353641000B6B8E
To: ;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:18:02] WARNING[10572] chan_sip.c: Retransmission timeout reached on transmission 0201C1A90C81400000000010@192.168.1.11 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

Configurare VoIP cu TLS si SRTP

danpopa — GMT

Salut.
Am incercat sa configurez serverul cu ghidul luat de aici https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
Dupa configurarile facute ca in tutorialul de mai sus, telefonul (Blink) nu se mai poate inregistra la server.
Cum as putea rezolva problema? Unde gresesc?
Multumesc!

Configurare Linksys Spa 2102 prin RDS(pppoe cu parola)

edmonddante — GMT

Buna ziua tuturor! Am o problema cu un router cu adaptor pentru telefon,Linksys Spa 2102.Nu reusesc sa ajung sa-i fac configurarea,practic,nu apuc sa o vad. Sunt in RDS si am legatura prin finra optica prin pppoe. Respectand instructiunile si facand legaturile aferente,in momentul cand pun pe browser adresa routerului nu primesc nimic inapoi fiindca internetul nu mai merge !Asa ca,nu am ce configurare sa fac .Cum s-ar putea face legatura prin acest router la internet? Am luat legatura cu cei de la RDS si mi-au zis ca e treaba mea,nu e echipamentul lor,nu-i intereseaza.Probabil ca o solutie exista,din pacate,nu o stiu. Daca stie cineva i-as ramane profund indatorat .Multumesc anticipat !

SPA 9000

turgay — GMT

Buna ziua

La un SPA 9000 pentru PSTN acces pot folosii in loc de SPA 400 modelul SPA 3102? Cel din urma mi se pare mai accesibil.

Multumesc

Aastra 55i + fxo Gateway

skunky — GMT

Salut,
Este oare posibil a ruta apelurile ce vin pe o linie a echipamentului fxo ( audiocodes mp118 ) direct catre telefonul Aastra 55i ?
Toate acestea fara un pbx intermediar.

Tone disconnect

ghostbit — GMT

Salut,

Am achizitionat de la voi un Grandstream GXW4108 si il folosesc impreuna cu un server de Asterisk.
Intern, totul functioneaza perfect: call setup rapid, calitate foarte buna, fara distorsiuni...etc.

In extern insa... am mici intreruperi (nu se pune problema de latente pe echipamentele interne de retea), iar sunetul (directia Rx) este variabil. Audio Gain-ul de pe gateway nu ma ajuta foarte mult.
Din experienta dvs... pe ce setari (gateway) ar trebui sa ma concentrez astfel incat sa rezolv aceste probleme.
Folosesc 5 linii de Romtelecom si 3 de Vodafone.

Multumesc.

Solutie VOIP

ghostbit — GMT

Salut,

Vreau sa realizez o infrastructura VOIP folosind ca PBX un server ce ruleaza Asterisk/FreePBX. Am 12 linii PSTN.
Doresc sa-mi comunicati o solutie hardware 'cost-effective' (SIP gateway cu interfete FXO), compatibila cu Asterisk (eventual cu documentatie ce specifica concret modul de conectare al echipamentelor la serverul Asterisk).
Multumesc

Sunt interesat in colaboarare

alpin — GMT

Salut, sunt interesat in colaborare penru o centrala asterisk. Am cateva modifiari de facut dar si doresc colaborare pe temen lung. Tipul care s-a ocupat pana acum nu mai are timp si pentru cei mici sau nu mai stie ce sa-i faca. Spun asta deoarece inca de la inceput eu nu am vazut apelurile si am asteptat 3 luni ca sa faca ceva. Cred ca este indeajuns. Este cineva interesat?
Astept mesaj aici.
Multumesc frumos

Gateway FXO 2 porturi compatibil Asterisk

paul — GMT

Ce gateway FXO, cu 2 pana la 4 porturi FXO, stiti ca merge bine cu Asterisk?

Am incercat pana acum un Welltech 3804A si un Audiocodes MP 114 FXO. Cu ambele am 2 probleme mari:
In primul rand nu functioneaza full duplex. Cand vorbeste una din parti, cealalta nu poate vorbi.
In al doilea rand nu functioneaza Caller ID.

Stiti un device care se mearga perfect cu Asterisk, ca nu as vrea sa mai cumpar inca un device inutil?

Zone si Trasee Montane

Nini — GMT

[b]19/Mai/2011[/b]

[list]8 [url=http://gis.modulo.ro/hiking/zoneMontane.html]zone montane[/url][/list][code]Piatra Mare
Muntii Capatanii
Fagaras
Cozia
Ceahlau
Bucegi
Ciucas
Piatra Craiului[/code][list]58 [url=http://gis.modulo.ro/OsmHiking/traseeMontane.html]trasee montane[/url][/list]

Adaugarea unui nou traseu

Nini — GMT

In [url=http://gis.modulo.ro/hiking/definireaTraseelorMontane.html]aceasta pagina[/url] gasiti un tutorial care descrie pasii necesari introducerii unui nou traseu in [url=http://www.openstreetmap.org]OpenStreetMap[/url] (OSM).

[b]Observatii[/b]:
[list]Daca nu aveti cont in OSM este necesar sa va creati unul, procedura foarte simpla si la indemana oricui.[/list]
[list]Intrebari generale despre OSM pot fi adresate pe [url=http://lists.openstreetmap.org/listinfo/talk-ro]lista de discutii[/url] a comunitatii OSM-RO[/list]
Dupa ce traseul va fi introdus in OSM acesta va aparea in [url=http://gis.modulo.ro]site[/url] la primul update.

[b]Atentie:[/b] daca traseul face parte dintr-o [url=http://gis.modulo.ro/hiking/zoneMontane.html]zona montana[/url] care nu este definita acesta nu va fi afisat. In astfel de cazuri va rugam sa ne anuntati (pe forum sau mail) despre ce zona montana este vorba pentru a putea sa o includem in baza de date.

Daca aveti intrebari sau nelamuriri despre modalitatea de adaugare a unui traseu va rugam sa le postati aici.

Buton Click2Call in aplicatii VBA - exemplu pentru Excel si FreePBX

Nini — GMT

Daca trebuie sa construiti o aplicatie in care sa exite un buton de tip "Click2Call" putem sa folositi fragmentele de cod de mai jos - pe baza lor se poate face o integrare a unei aplicatii scrisa in VBA cu Asterisk (in cazul nostru Excel si Asterisk cu FreePBX).

In mod cert exista si alte abordari (de ex. folosirea [url=http://www.voip-info.org/wiki/view/Asterisk+manager+API]AMI[/url] in loc de [url=http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out]CallFiles[/url]) si de aceea asteptam sugestii sau comentarii pe marginea celor prezentate in acest topic.

[b][i]Nota: Trebuie luate precautiile necesare pentru evitarea folosirii abuzive a comenzilor de sistem. Codurile mentionate sunt date cu titlu de exemplu si trebuie tratate ca atare.[/i][/b]

1. Cod VBA asociat unui buton din Excel (client-side)
[code]Private Sub CommandButton1_Click()
Set HttpReq = CreateObject("Microsoft.XMLHTTP")
HttpReq.Open "GET", "http://tenora.your-domain.ro/click2call.php?called=" & Me.Cells(3, 1) & "&calling=" & Me.Cells(3, 2), False
HttpReq.send
End Sub[/code]
2. Cod PHP (server-side)
[code] $called = $_GET['called'];
$calling= $_GET['calling'];
echo '

';

 $cmd = system('/etc/asterisk/tenora/tools/gen_call_to_context Local/'.$called.'@from-internal from-internal '.$calling, $retval);

 echo '

 Last line of the output: ' . $cmd . '

 Return value: ' . $retval;

 echo '

';
?>[/code]
3. Script generare apel (server-side)
[code]echo "Calling $1 and connect to context $2 extension $3 priority 1"

CALL_FILE=/tmp/$(date "+%s").call
cat < $CALL_FILE
Channel: $1
MaxRetries: 0
Context: $2
Extension: $3
Priority: 1
EOF

chown asterisk.asterisk $CALL_FILE
mv $CALL_FILE /var/spool/asterisk/outgoing
[/code]

DAHDI - erori "Failed VPMADT032 reset. VPMADT032 is disabled."

Nini — GMT

In cazul in care observati in logurile de sistem erori de tipul
[code]wcte12xp 0000:02:08.0: Failed to load the firmware.
wcte12xp 0000:02:08.0: Failed VPMADT032 reset. VPMADT032 is disabled.[/code]
aceasta inseamna, cel mai probabil, ca firmware loader-ul pentru VPMADT032 (modulul de echo cancelling) nu este instalat.

In majoritatea cazurilor se intampla pe sistemele Trixbox 2.8, care nu distribuie acest modul insa Digium pune la dispozitie pachetul necesar pentru instalarea acestuia.

Instructiunile de instalare sunt sumarizate mai jos:

[list]Se identifica [url=http://packages.digium.com/trixbox/28/current/i386/RPMS/]de aici[/url] pachetul care sa corespunda sistemului dvs. Cel mai probabil primul din lista este cel care va trebuie[/list]
[list]Se transfera pachetul respectiv pe sistemul dvs.[/list][code]cd /tmp
wget http://packages.digium.com/trixbox/28/current/i386/RPMS/kmod-dahdi-linux-fwload-vpmadt032-2.3.0.1-1_trixbox28.2.6.18_164.11.1.el5.i686.rpm[/code]
[list]Se instaleaza pachetul utilizand managerul de pachete RPM[/list][code]rpm -Uvh kmod-dahdi-linux-fwload-vpmadt032-2.3.0.1-1_trixbox28.2.6.18_164.11.1.el5.i686.rpm
[/code]
[list]Dupa aceasta se procedeaza la restart-ul DAHDI[/list]
[code]amportal stop && /etc/init.d/dahdi restart && amportal start[/code]

Dupa aceasta operatie ar trebui sa regasiti in loguri informatii asemanatoare cu cele de mai jos:
[code]dahdi_vpmadt032_loader: module license 'Digium Commercial' taints kernel.
dahdi_vpmadt032_loader: no version for "vpmadtreg_unregister" found: kernel tainted.
wcte12xp 0000:02:08.0: VPM present and operational (Firmware version 120)
...
wcte12xp 0000:02:08.0: Booting VPMADT032
wcte12xp 0000:02:08.0: VPM present and operational (Firmware version 120)[/code]

Daca aveti probleme in continuare va rog sa postati aici si vom incerca sa gasim, impreuna, o solutie.

Ioan.

SPA 3102 si RCS - RDS 2

andrei — GMT

Am si eu o problema cu spa 3102 in combinatie cu rdstel. SPA 3102 e legat de asterisk. Apelurile de pe rds merg catre o extensie. Dupa ce raspund la apel pe extensia respectiva daca cel care a sunat inchide imi vine ton de ocupat pe linie si daca astept suficient de mult imi vine tiutiul de la rds sa inchid telefonul. Din cate imi dau seama spa-ul nu detecteaza tonul de ocupat desi am incercat diverse variante si logic ar fi sa mearga cu 450@-10; 10(0.17/0.17/1) la Disconnect Tone cod pe care l-am gasit si pe un site cu coduri pt. spa dar l-am calculat si eu pe o monstra inregistrata a tonului de ocupat. S-a mai lovit cineva de problema asta, cum pot sa fac sa se inchida totusi conexiunea ?

Multumesc