timaca Posted April 3, 2014 Report Share Posted April 3, 2014 Hello, I can't register a spa112 with my snomone pbxI don't know what is the problem.Here a log from the server: REGISTER sip:100.64.0.49 SIP/2.0Via: SIP/2.0/UDP 192.168.215.40:5060;branch=z9hG4bK-cedabad4From: "556" <sip:556@100.64.0.49>;tag=50af8cc356fe346o0To: "556" <sip:556@100.64.0.49>Call-ID: 441e8cfa-daaa8395@192.168.215.40CSeq: 1467 REGISTERMax-Forwards: 70Contact: "556" <sip:556@192.168.215.40:5060>;expires=3600User-Agent: Cisco/SPA112-1.3.3(015)P-Station-Name: ;mac=34dbfd5c1389; display=""; sn=CCQ174101ZI--SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.215.40:5060;branch=z9hG4bK-cedabad4From: "556" <sip:556@100.64.0.49>;tag=50af8cc356fe346o0To: "556" <sip:556@100.64.0.49>;tag=0a6a3267ecCall-ID: 441e8cfa-daaa8395@192.168.215.40CSeq: 1467 REGISTER Thanks Quote Link to comment Share on other sites More sharing options...
Vodia support Posted April 6, 2014 Report Share Posted April 6, 2014 is the device on a WAN? looks like it's missing the domain, unless you add the 100.x.x.x as the primary domain name it should register. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted April 7, 2014 Report Share Posted April 7, 2014 If you "REGISTER sip:100.64.0.49 SIP/2.0" the PBX must have a domain "100.64.0.49" or a domain with the name "localhost" in it. Otherwise it will not match the domain. Quote Link to comment Share on other sites More sharing options...
Guest timaca Posted April 8, 2014 Report Share Posted April 8, 2014 thank you for your reponses. I actually finished after many tests by putting the domain name in the "proxy" field and 100.64.0.49 in the "outbound proxy". The SPA112 is now registered. I now have other worries, line 1 on which a fax is plugged in, works once three, either in transmission or reception. there is an interrupt after a few tens of seconds. On line 2 I have equipment to free mail. when making a call, it cuts after exactly one minute. My trunk provider doesn't support T38. Here the traces of spa112: (incoming fax) SIP_regTsEventProc(event: 32) SIP_tsClientEventProc(event: 3) SIP_tsClientEventProc(event: 9) SIP_tsClientEventProc(event: 9) SIP_tsCreateClient(), 1779, uiTmrF=1600, SIP_TMR_F_INIT=1600 SIP_tsClientEventProc(event: 28) SIP_regTsEventProc(event: 28) SIP_regTsEventProc(event: 32) CC_eventProc(), event: CC_EV_SIG_CALL_ARRIVED(0x31), lid: 0, par: 0, par2: 0x407b8900 AUD_ccEventProc: event 49 vid 0 par 0x0 par2 0x407b8900 CC_lineBusy() getCodecList line 0x22aae8 call 0x22aaec clRemote: 0x407b8990 bInbound 1 pconly: 128 ====== Remote Codec num 3 ====== 08136================================= ====== Local codec num 1 ====== 0================================ [AUD]Get UCH node for AUD_LINE 0 0. uchAllocateNode(), Node 0 allocated ret=0 [AUD]UCH node 0 allocated to AUD_LINE 0. uchConnectEpToNode(), connecting EP FXS 1 to node 0 uchEnableNode(), Node 0 enbaled ret=0 CC_eventProc(), inf.strName = CC_eventProc(), inf.strPhone = 0954854540 callEventProcTable[0] is cepIdleProc cepIdleProc(lid=0, call=0x22aaec, event=18(CC_EV_USR_ACCEPTCALL), par=0, par2=0x407b8900) cepIdleProc(), lid=0 cepIdleProc(), line->sigProc(CC_CMD_ACCEPT) cepIdleProc(), call->cinf.bAutoAnswer = 0 NEW_CALL_STATE(), call 0: old state = CC_CST_IDLE, new state CC_CST_RINGING CID_initGen 8 [0]CID:CID_initGen() >>> offhook 0 delay 2200 phone 0954854540 name SLIC_startRing state 0 ts 0x25a644on 2000 off 4000 len 60000 [0]Ring cad event 0 pol 0 RTP_nextMediaPort(), port = 16414 RTP_nextMediaPort(), rc=16412 AUD_allocCallObj() call(0x25ce38) [0:0]AUD ALLOC CALL (port=16412) +++++ SIP_lineCcCmdProc CC_CMD_ACCEPT AUD_startRtpRx [AUD]AUD_startRtpRx(0x25ce38) lid 0 Local loopback mode: None. Type: None. Remote loopback mode: None. Type None. RTP channel setup: udp_no_checksum 0, sysmmetric_rtp 0, tos 0xb8, cos 6, mlb 0. uchConnectEpToNode(), connecting EP VoIP 0 to node 0 uchEnableEchoCan(), lid 0 EP 2 disable uchEnableModemCall() Modem call state(1) not change UCH sync parameter hold off time is 70 uchSetGTD(), Disable GTD for FXS 1 uchSetGTD() GTD state not change uchSetDTMFMute(), ENABLE cordless_start_rtp(), chan:0 remote ip:(null) port:0 local:16412 rx:1 ipt:0 ptime:30 bInMdmPasstru:0 Starting Rx only RTP. Socket 14 bound to port 16412. Remote IP/port: 0.0.0.0:0 Codec list from SDP (internal pt): 0 134 136 Rx payload list: PCMU/8000(0) NSE/8000(100) encaprtp/8000(112) set RTP_SESSION_OPT_DTMF VAD = 0 RTP configuration: audio_mode RTP_MODE_REC_ONLY, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0 Codec: duration 30, rx_pt_event 101, tx_pt_event 101, tx_pt 0 rx[0] 0 PCMU/8000, rx[1] 100 NSE/8000, rx[2] 112 encaprtp/8000 rx[3] -1 , rx[4] -1 , rx[5] -1 Jib: max 180ms, min 60ms, adapt 1 RTP Channel 0 is virgin: 1. #### rtp seq number is 4975 Set QoS succeed RTP session 0 started [AUD]RTP Rx Up SIP_tsClientEventProc(event: 3) SIP_tsCreateClient(), 1779, uiTmrF=1600, SIP_TMR_F_INIT=1600 SIP_tsClientEventProc(event: 28) SIP_regTsEventProc(event: 28) SIP_regTsEventProc(event: 32) SIP_tsClientEventProc(event: 3) [0]Ring cad event 1 pol 0 CID:OnHookTx Pol [0]CID CID_ST_POLREV_POST_DELAY uchDisplayCIDFSK(), EP 2 lid 0 buflen 97 overhead 60 SZ_MAX_USERDATA 200 offhook 0 uchDisplayCIDFSK(), FSK Caller ID standard is 0(bell 202) uchDisplayCIDFSK(), SeizeFreq 0x16 MarkFreq 0xc [0]CID Start DTMF/FSK, CID_ST_ACTIVE [0]Off Hook CC_eventProc(), event: CC_EV_USR_OFFHOOK(0x2), lid: 0, par: 0, par2: (nil) AUD_ccEventProc: event 2 vid 0 par 0x0 par2 0x0 sysstatus_set_led_status_payton(), led_id: 1, statusCode:7, systemEvent: 0x100095 callEventProcTable[5] is cepRingingProc cepRingingProc(lid=0, call=0x22aaec, event=11(CC_EV_USR_ANSWER), par=0, par2=(nil)) NEW_CALL_STATE(), call 0: old state = CC_CST_RINGING, new state CC_CST_ANSWERING SLIC_stopRing [0]Ring cad event 2 pol 0 SLIC_stopRing SLIC_stopTone uchStopCTI(), Stop CTI lid 0 EP 2,ret=0 [0]CID interrupted uchAppCb(), Event 49 received EP 2 lid 0 receive CH_ASYNC_CIT_TRANSMITTED uchStopVoipTone(), Stop Voip Tone EP 3 SIP_sessDlgEventProc: event: 43(SIP_EV_DLG_CONNECTED), ucState: 0 CC_eventProc(), event: CC_EV_SIG_CALL_CONNECTED(0x2C), lid: 0, par: 10, par2: (nil) AUD_ccEventProc: event 44 vid 0 par 0xa par2 0x0 callEventProcTable[7] is cepAnsweringProc cepAnsweringProc(lid=0, call=0x22aaec, event=44(CC_EV_SIG_CALL_CONNECTED), par=10, par2=(nil)) CC:Connected NEW_CALL_STATE(), call 0: old state = CC_CST_ANSWERING, new state CC_CST_CONNECTED SLIC_stopRing SLIC_stopRing SLIC_stopTone +++++ SIP_sessDlgEventProc SIP_SST_ACCEPTED SIP_EV_DLG_CONNECTED START_RTPTX bSendRtpTxToProxy: ip: 100.64.0.49 [AUD]AUD_startRtpTx(0x25ce38, 0, 100.64.0.49, 61012, 30) Local loopback mode: None. Type: None. Remote loopback mode: None. Type None. Already has a RTP channel. Already has a RTP channel. cordless_start_rtp(), chan:0 remote ip:100.64.0.49 port:61012 local:16412 rx:0 ipt:0 ptime:30 bInMdmPasstru:0 Going from Rx only to bi-directional. Old remote IP/port: 0.0.0.0:0 Remote IP/port: 100.64.0.49:61012 Codec list from SDP (internal pt): 0 134 136 Rx payload list: PCMU/8000(0) NSE/8000(100) encaprtp/8000(112) set RTP_SESSION_OPT_DTMF VAD = 0 RTP configuration: audio_mode RTP_MODE_ACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0 Codec: duration 30, rx_pt_event 101, tx_pt_event 101, tx_pt 0 rx[0] 0 PCMU/8000, rx[1] 100 NSE/8000, rx[2] 112 encaprtp/8000 rx[3] -1 , rx[4] -1 , rx[5] -1 Jib: max 180ms, min 60ms, adapt 1 RTP Channel 0 is virgin: 0. Just need updating. RTP session 0 updated [AUD]RTP Tx Up [AUD]AUD_startRtcpTx(0x25ce38) cordless_start_rtcp(), chan:0 remote ip:100.64.0.49 port:61013 intvl:0 Socket 19 bound to RTCP port 16413. CNAME 556@192.168.215.40 NAME "SPA112-Fax-Gavotte" TOOL Cisco/SPA112-1.3.3(015) Starting RTCP session on channel 0. Interval 0. Rx only. RTCP session started on RTP channel 0. [AUD]RTCP Up SIP_tsClientEventProc(event: 9) SIP_tsClientEventProc(event: 9) uchAppCb(), Event 46 received EP 2 lid 0 fax event detected tone type: 8 at line 0 1 fax event enable CED 1024, CNG 64 complete dialog 1 lid: 0 ep: 2 uchAppCb(), Event 46 received EP 2 lid 0 fax event detected tone type: 8 at line 0 1 fax event enable CED 1024, CNG 64 complete dialog 1 lid: 0 ep: 2 ANS Tone Phase Rev detected Going into fax passthrough mode uchEnableEchoCan(), lid 0 EP 2 disable uchEnableModemCall() Modem call state(1) not change VAD = 0 RTP configuration: audio_mode RTP_MODE_ACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0 Codec: duration 30, rx_pt_event 101, tx_pt_event 101, tx_pt 0 rx[0] 0 PCMU/8000, rx[1] 100 NSE/8000, rx[2] 112 encaprtp/8000 rx[3] -1 , rx[4] -1 , rx[5] -1 Jib: max 1000ms, min 60ms, adapt 0 cordless_enableFaxPassthroughWithCodec *** rtp_sessions_update() ok: 0 cordless_enableFaxPassthroughWithCodec *** rtp_sessions_update() ok: 0 SIP_tsCreateClient(), 1779, uiTmrF=1600, SIP_TMR_F_INIT=1600 SIP_tsClientEventProc(event: 28) SIP_regTsEventProc(event: 28) SIP_regTsEventProc(event: 32) SIP_tsClientEventProc(event: 3) SIP_tsCreateClient(), 1779, uiTmrF=1600, SIP_TMR_F_INIT=1600 SIP_tsClientEventProc(event: 28) SIP_regTsEventProc(event: 28) SIP_regTsEventProc(event: 32) SIP_tsClientEventProc(event: 3) SIP_tsClientEventProc(event: 9) SIP_tsClientEventProc(event: 9) httpd_handle_request(), request method = 1 httpd_handle_request(), request path = /admin/voice/ httpd_handle_request(), pswlReq->ubType = 0 Requesting call statistics... RTP TX stats updated for channel 0 RTP RX stats updated for channel 0 Call statistics updated. SIP_tsCreateClient(), 1779, uiTmrF=1600, SIP_TMR_F_INIT=1600 SIP_tsClientEventProc(event: 28) SIP_regTsEventProc(event: 28) SIP_regTsEventProc(event: 32) SIP_tsClientEventProc(event: 3) SIP_tsCreateClient(), 1779, uiTmrF=1600, SIP_TMR_F_INIT=1600 SIP_tsClientEventProc(event: 28) SIP_regTsEventProc(event: 28) SIP_regTsEventProc(event: 32) SIP_tsClientEventProc(event: 3) SIP_tsClientEventProc(event: 9) SIP_tsClientEventProc(event: 9) SIP_sessTsEventProc(event:27) Requesting call statistics... RTP TX stats updated for channel 0 RTP RX stats updated for channel 0 Call statistics updated. SIP_sessDlgEventProc: event: 45(SIP_EV_DLG_BYED), ucState: 3 CC_eventProc(), event: CC_EV_SIG_CALL_ENDED(0x34), lid: 0, par: 10, par2: (nil) AUD_ccEventProc: event 52 vid 0 par 0xa par2 0x0 callEventProcTable[8] is cepConnectedProc cepConnectedProc(lid=0, call=0x22aaec, event=52(CC_EV_SIG_CALL_ENDED), par=10, par2=(nil)) CC:Ended NEW_CALL_STATE(), call 0: old state = CC_CST_CONNECTED, new state CC_CST_INVALID CPC statr timer CC_EV_TMR_INVALID SLIC_stopRing SLIC_stopRing SLIC_stopTone SIP_releaseAudioResources() entered ################!!!!!!!!!!!!!!!!! Requesting call statistics... RTP TX stats updated for channel 0 RTP RX stats updated for channel 0 Call statistics updated. AUD_releaseCallObj() call(0x25ce38) [AUD]AUD_stopRtpTx(0x25ce38) cordless_stop_rtp_tx(), Channel 0. RTP channel 0 going from Bi-dir to Rx. RTP configuration: audio_mode RTP_MODE_REC_ONLY, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0 Codec: duration 30, rx_pt_event 101, tx_pt_event 101, tx_pt 0 rx[0] 0 PCMU/8000, rx[1] 100 NSE/8000, rx[2] 112 encaprtp/8000 rx[3] -1 , rx[4] -1 , rx[5] -1 Jib: max 1000ms, min 60ms, adapt 0 RTP channel 0 is now Rx. [AUD]RTP Tx Down [AUD]AUD_stopRtpRx(0x25ce38) cordless_stop_rtp_rx(), Channel 0. RTP channel 0 going from Rx to Idle. RTP configuration: audio_mode RTP_MODE_INACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0 Codec: duration 30, rx_pt_event 101, tx_pt_event 101, tx_pt 0 rx[0] 0 PCMU/8000, rx[1] 100 NSE/8000, rx[2] 112 encaprtp/8000 rx[3] -1 , rx[4] -1 , rx[5] -1 Jib: max 1000ms, min 60ms, adapt 0 RTP channel 0 is now Idle. [AUD]RTP Down [AUD]AUD_releaseRtp(0x25ce38) cordless_stop_rtp(), releasing RTP channel:0 cordless_stop_rtp(), RTP session 0 stopped succussfully uchRelChanAndEP(0, 3) uchDisconnectEpFromNode(), disconnecting EP VoIP 0 from node 0 [AUD]RTP channel released [0:0]AUD Rel Call SIP_releaseAudioResources(), CC_lineIsIdle(0)=0, gAudLine[0].bIvr=0, AUD_relUchNode???????????? SIP_releaseAudioResources() exit ################!!!!!!!!!!!!!!!!! ##### RTP_SEQ_NUM_EVT 6268 callEventProcTable[6] is cepInvalidProc cepInvalidProc(lid=0, call=0x22aaec, event=31(CC_EV_TMR_INVALID), par=0, par2=(nil)) CPC SLIC_SET_OPEN_STATE on line 0 SLIC_stopRing SLIC_stopRing SLIC_stopTone callEventProcTable[6] is cepInvalidProc cepInvalidProc(lid=0, call=0x22aaec, event=39(CC_EV_TMR_CPC), par=0, par2=(nil)) CPC go to CC_CST_IDLE line 0 NEW_CALL_STATE(), call 0: old state = CC_CST_INVALID, new state CC_CST_IDLE [AUD]Release UCH node for AUD_LINE 0. uchDisableNode(), Node 0 released ret=0 [AUD]UCH node 0 freed. callEventProcTable[0] is cepIdleProc cepIdleProc(lid=0, call=0x22aaec, event=9(CC_EV_USR_SEIZURE), par=0, par2=(nil)) cepIdleProc(), lid=0 cepIdleProc(), pname=renaissance cepIdleProc(), SYS_NOREG_CALL(0)=0, SIP_REGISTER_OK(0)=1 [AUD]Get UCH node for AUD_LINE 0 0. uchAllocateNode(), Node 0 allocated ret=0 [AUD]UCH node 0 allocated to AUD_LINE 0. uchConnectEpToNode(), connecting EP FXS 1 to node 0 uchEnableNode(), Node 0 enbaled ret=0 NEW_CALL_STATE(), call 0: old state = CC_CST_IDLE, new state CC_CST_DIALING SLIC_stopRing SLIC_startTone 1 [0]On Hook CC_eventProc(), event: CC_EV_USR_ONHOOK(0x1), lid: 0, par: 0, par2: (nil) AUD_ccEventProc: event 1 vid 0 par 0x0 par2 0x0 sysstatus_set_led_status_payton(), led_id: 1, statusCode:1, systemEvent: 0x100093 callEventProcTable[1] is cepDialingProc cepDialingProc(lid=0, call=0x22aaec, event=1(CC_EV_USR_ONHOOK), par=0, par2=(nil)) cepDialingProc(), event = 1(CC_EV_USR_ONHOOK) callEventProcTable[0] is cepIdleProc cepIdleProc(lid=0, call=0x22ad0c, event=1(CC_EV_USR_ONHOOK), par=0, par2=(nil)) cepIdleProc(), lid=0 [IVR_eventProc] evt 1 lid 0 callEventProcTable[1] is cepDialingProc cepDialingProc(lid=0, call=0x22aaec, event=10(CC_EV_USR_ENDCALL), par=0, par2=(nil)) cepDialingProc(), event = 10(CC_EV_USR_ENDCALL) NEW_CALL_STATE(), call 0: old state = CC_CST_DIALING, new state CC_CST_IDLE Set QoS succeed [AUD]Release UCH node for AUD_LINE 0. uchDisableNode(), Node 0 released ret=0 [AUD]UCH node 0 freed. SLIC_stopRing SLIC_stopTone ++++ retry query scaps ++++ retry query scaps +++ need tftp addr.. +++ need tftp addr.. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted April 9, 2014 Report Share Posted April 9, 2014 My trunk provider doesn't support T38. Here the traces of spa112: (incoming fax) Well honestly, then change the trunk provider. If you need to receive FAX, T.38 is a must-have. If you don't have it we all will go through a long, frustrating experience of FAX not going through randomly or not at all. We also have a SPA112 here, will add it to the provisioning process to make life easier for the next customers who set this device up. Quote Link to comment Share on other sites More sharing options...
timaca Posted April 9, 2014 Author Report Share Posted April 9, 2014 Well honestly, then change the trunk provider. If you need to receive FAX, T.38 is a must-have. If you don't have it we all will go through a long, frustrating experience of FAX not going through randomly or not at all. We also have a SPA112 here, will add it to the provisioning process to make life easier for the next customers who set this device up. I just made a request (again) to my service provider for the protocol "T38". In attentadnt, is there any solutions for reliable operation with g711? Traces I provided does not indicate a configuration problem ? thank you Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted April 9, 2014 Report Share Posted April 9, 2014 If you try G.711, make sure that you turn echo cancellation off on the SPA. Otherwise the echo canceler will kill the FAX signal. Fax was not designed for packet networks. It can deal with distortion and lots of analog problems, but if a whole piece (packet) gets lost, it is the end of the transmission. Nobody will change that for you in the year 2014 any more. Quote Link to comment Share on other sites More sharing options...
timaca Posted April 9, 2014 Author Report Share Posted April 9, 2014 If you try G.711, make sure that you turn echo cancellation off on the SPA. Otherwise the echo canceler will kill the FAX signal. Yes I've already turn off echo cancellation (without better results) Fax was not designed for packet networks. It can deal with distortion and lots of analog problems, but if a whole piece (packet) gets lost, it is the end of the transmission. Nobody will change that for you in the year 2014 any more. OK, and my trunk provider says: G711 protocol should give satisfactory results... (for me it's catastrophic) anyway, It doesn't explain the outgoing call failure on line 2.. yes ? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted April 10, 2014 Report Share Posted April 10, 2014 Well T.38 was invented for a reason. This protocol also has its problems, but I would say it is your best shot to get FAX transmitted. Some ATA also support HTTPS fax, which makes it really 100 % reliable; but the support for this is in the service provider world is homeopathic. Quote Link to comment Share on other sites More sharing options...
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