Re: [Asterisk-ES] Problema con FAX, FXS, TDM400, Proveedor SIP + T.38
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Fernando Villares
Nov 12, 2012, 1:36:26 PM11/12/12
to aster...@googlegroups.com
1ero 14400 es un poco optimista...bajale a 9600 o 4800 de max y 2400 de min esta bien...2ndo, estas segurisimo que tu provider soporta t38 de verdad...prueba sencilla...gateway o ata cisco 112 o grandstream 502 o 701 o un audiocodes m202 fxs...que si soportan t38, poneselo a tu provider como extension y proba desde la pstn con el fax tradicional mandar a la did del sip...si pasa bien sin problemas listo ay sabes que tu provider soporta...ahora si se corta podes empezar a buscar desde ahi...que ya sabes que no funciona empiricamente el proveedor sip
2012/11/8 bitxit0 <david.jesu...@gmail.com>
Buenas tardes, tengo una serie de dudas que por más que googleo no consigo solucionar, el escenario que quiero conseguir es el siguiente :
Máquina de FAX tradicional ---> FXS de mi TDM400 ---> Asterisk + FFA ---> Proveedor SIP con soporte T.38 ---> PSTN ---> Máquina de FAX tradicional
Ahora mismo estoy atascado en intentar enviar un FAX, creo que lo tengo todo bien configurado, en el CLI el comando fax show capabilities
devuelve:
CLI> fax show capabilities
Registered FAX Technology Modules:
Type : DIGIUM
Description : Digium FAX Driver
Capabilities : SEND RECEIVE T.38 G.711 MULTI-DOC
1 registered modules
CLI>
CLI> fax show stats
FAX Statistics:
---------------
Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 0
Completed FAXes : 0
Failed FAXes : 0
Digium G.711
Licensed Channels : 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0
Digium T.38
Licensed Channels : 1
Max Concurrent : 0
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0
pulsar*CLI>
Primera duda : en Capabilities no deberia aparecer GATEWAY ?¿?¿?
Cuando intento enviar un FAX me sucede lo siguiente :
CLI>
[Nov 8 18:55:31] DEBUG[2306]: chan_dahdi.c:9789 do_monitor: Monitor doohicky got event Ring/Answered on channel 3
[Nov 8 18:55:31] DEBUG[2306]: dsp.c:473 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Nov 8 18:55:31] DEBUG[2306]: dsp.c:473 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Nov 8 18:55:31] DEBUG[2306]: devicestate.c:442 devstate_event: device 'DAHDI/3-1' state '2'
[Nov 8 18:55:31] DEBUG[2184]: app_queue.c:1082 handle_statechange: Device 'DAHDI/3-1' changed to state '2' (In use) but we don't care because they're not a member of any queue.
-- Starting simple switch on 'DAHDI/3-1'
[Nov 8 18:55:32] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: 9 on DAHDI/3-1
[Nov 8 18:55:32] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned '9' (57), timeout = 0
[Nov 8 18:55:32] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: X on DAHDI/3-1
[Nov 8 18:55:32] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned 'X' (55), timeout = 0
[Nov 8 18:55:32] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: X on DAHDI/3-1
[Nov 8 18:55:32] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned 'X' (49), timeout = 0
[Nov 8 18:55:32] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: X on DAHDI/3-1
[Nov 8 18:55:32] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned 'X' (51), timeout = 0
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: X on DAHDI/3-1
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned 'X' (57), timeout = 0
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: X on DAHDI/3-1
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned 'X' (53), timeout = 0
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: X on DAHDI/3-1
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned 'X' (49), timeout = 0
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: X on DAHDI/3-1
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned 'X' (53), timeout = 0
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:5687 dahdi_handle_dtmfup: DTMF digit: X on DAHDI/3-1
[Nov 8 18:55:33] DEBUG[2868]: chan_dahdi.c:8156 ss_thread: waitfordigit returned 'X' (53), timeout = 0
[Nov 8 18:55:37] DEBUG[1948]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for DAHDI - 3
[Nov 8 18:55:37] DEBUG[1948]: devicestate.c:462 do_state_change: Changing state for DAHDI/3 - state 2 (In use)
[Nov 8 18:55:37] DEBUG[1948]: devicestate.c:442 devstate_event: device 'DAHDI/3' state '2'
[Nov 8 18:55:37] DEBUG[2184]: app_queue.c:1082 handle_statechange: Device 'DAHDI/3' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Nov 8 18:55:37] DEBUG[2868]: chan_dahdi.c:2688 dahdi_enable_ec: Enabled echo cancellation on channel 3
[Nov 8 18:55:37] DEBUG[2868]: pbx.c:3692 pbx_extension_helper: Launching 'NoOp'
-- Executing [9XXXXXXXX@outboundfax:1] NoOp("DAHDI/3-1", "") in new stack
[Nov 8 18:55:37] DEBUG[2868]: pbx.c:3692 pbx_extension_helper: Launching 'Set'
-- Executing [9XXXXXXXX@outboundfax:2] Set("DAHDI/3-1", "FAXOPT(gateway)=yes") in new stack
[Nov 8 18:55:37] DEBUG[2868]: res_fax.c:2632 acf_faxopt_write: channel 'DAHDI/3-1' setting FAXOPT(gateway) to 'yes'
[Nov 8 18:55:37] WARNING[2868]: res_fax.c:2660 acf_faxopt_write: channel 'DAHDI/3-1' set FAXOPT(gateway) to 'yes' is unhandled!
[Nov 8 18:55:37] DEBUG[2868]: pbx.c:3692 pbx_extension_helper: Launching 'Dial'
-- Executing [9XXXXXXXX@outboundfax:3] Dial("DAHDI/3-1", "SIP/ProveedorSIP/00349XXXXXXXX,20") in new stack
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:23330 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw)
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:7463 sip_alloc: Allocating new SIP dialog for 068b6baa4e6b0b84...@127.0.0.1 - INVITE (With RTP)
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:5141 do_setnat: Setting NAT on RTP to On
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:5149 do_setnat: Setting NAT on UDPTL to On
[Nov 8 18:55:37] DEBUG[2868]: acl.c:506 ast_ouraddrfor: Found IP address for this socket
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:3810 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.2.127:5060
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:6824 sip_new: *** Our native formats are 0x2 (gsm)
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:6825 sip_new: *** Joint capabilities are 0x0 (nothing)
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:6826 sip_new: *** Our capabilities are 0xa (gsm|alaw)
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:6827 sip_new: *** AST_CODEC_CHOOSE formats are 0x2 (gsm)
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:6829 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw)
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:6857 sip_new: This channel will not be able to handle video.
[Nov 8 18:55:37] DEBUG[2868]: rtp.c:2225 ast_rtp_make_compatible: Channel 'DAHDI/3-1' has no RTP, not doing anything
[Nov 8 18:55:37] DEBUG[2868]: channel.c:4503 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Nov 8 18:55:37] DEBUG[2868]: channel.c:4503 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Nov 8 18:55:37] DEBUG[2868]: channel.c:4503 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Nov 8 18:55:37] DEBUG[2868]: channel.c:4503 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Nov 8 18:55:37] DEBUG[2868]: channel.c:4503 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Nov 8 18:55:37] DEBUG[2868]: channel.c:4503 ast_channel_inherit_variables: Not copying variable TRANSFERCAPABILITY.
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:5567 sip_call: Outgoing Call for 00349XXXXXXXX
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:5783 update_call_counter: Updating call counter for outgoing call
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:10307 add_sdp: ** Our capability: 0xa (gsm|alaw) Video flag: False Text flag: False
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:10308 add_sdp: ** Our prefcodec: 0x4 (ulaw)
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:10419 add_sdp: -- Done with adding codecs to SDP
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:10555 add_sdp: Done building SDP. Settling with this capability: 0xa (gsm|alaw)
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:3402 initialize_initreq: Initializing initreq for method INVITE - callid 7b8b2ca875e826c4...@192.168.2.127
[Nov 8 18:55:37] DEBUG[2868]: chan_sip.c:3689 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for XXX.XXX.XXX.XXX:5060
-- Called ProveedorSIP/00349XXXXXXXX
[Nov 8 18:55:37] DEBUG[2868]: channel.c:3881 set_format: Set channel SIP/ProveedorSIP-00000005 to read format slin
[Nov 8 18:55:37] DEBUG[2868]: channel.c:3881 set_format: Set channel DAHDI/3-1 to write format slin
[Nov 8 18:55:37] DEBUG[2868]: channel.c:3881 set_format: Set channel DAHDI/3-1 to read format slin
[Nov 8 18:55:37] DEBUG[2868]: channel.c:3881 set_format: Set channel SIP/ProveedorSIP-00000005 to write format slin
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:4188 __sip_ack: Acked pending invite 102
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:4225 __sip_ack: Stopping retransmission on '7b8b2ca875e826c4...@192.168.2.127' of Request 102: Match Found
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:17792 handle_response_invite: SIP response 401 to standard invite
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:3689 __sip_xmit: Trying to put 'ACK sip:003' onto UDP socket destined for XXX.XXX.XXX.XXX:5060
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:17231 do_proxy_auth: Auth attempt 1 on INVITE
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:10307 add_sdp: ** Our capability: 0xa (gsm|alaw) Video flag: False Text flag: False
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:10308 add_sdp: ** Our prefcodec: 0x4 (ulaw)
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:10419 add_sdp: -- Done with adding codecs to SDP
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:10555 add_sdp: Done building SDP. Settling with this capability: 0xa (gsm|alaw)
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:3689 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for XXX.XXX.XXX.XXX:5060
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:4266 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7b8b2ca875e826c4...@192.168.2.127' Request 103: Found
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:17792 handle_response_invite: SIP response 100 to standard invite
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:4266 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7b8b2ca875e826c4...@192.168.2.127' Request 103: Found
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:17792 handle_response_invite: SIP response 183 to standard invite
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP o=login 1352397322 1352397322 IN IP4 XXX.XXX.XXX.XXX... UNSUPPORTED.
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED.
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP c=IN IP4 XXX.XXX.XXX.XXX... OK.
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8532 process_sdp: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8532 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8532 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK.
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8736 process_sdp: We're settling with these formats: 0x2 (gsm)
[Nov 8 18:55:37] DEBUG[2276]: chan_sip.c:8741 process_sdp: We have an owner, now see if we need to change this call
-- SIP/ProveedorSIP-00000005 is making progress passing it to DAHDI/3-1
[Nov 8 18:55:37] DEBUG[2868]: chan_dahdi.c:7254 dahdi_indicate: Requested indication 14 on channel DAHDI/3-1
[Nov 8 18:55:37] DEBUG[2868]: chan_dahdi.c:7371 dahdi_indicate: Received AST_CONTROL_PROGRESS on DAHDI/3-1
[Nov 8 18:55:37] DEBUG[2868]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to gsm
[Nov 8 18:55:37] DEBUG[2868]: rtp.c:3904 ast_rtp_write: Created smoother: format: 2 ms: 20 len: 33
pulsar*CLI>
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:4188 __sip_ack: Acked pending invite 103
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:4225 __sip_ack: Stopping retransmission on '7b8b2ca875e826c4...@192.168.2.127' of Request 103: Match Found
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:17792 handle_response_invite: SIP response 200 to standard invite
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP o=login 1352397333 1352397333 IN IP4 XXX.XXX.XXX.XXX... UNSUPPORTED.
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED.
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP c=IN IP4 XXX.XXX.XXX.XXX... OK.
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8368 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8532 process_sdp: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8532 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8532 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK.
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8736 process_sdp: We're settling with these formats: 0x2 (gsm)
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:8741 process_sdp: We have an owner, now see if we need to change this call
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:5783 update_call_counter: Updating call counter for outgoing call
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:12699 build_route: build_route: Contact hop: sip:00349X...@XXX.XXX.XXX.XXX:5060
[Nov 8 18:55:48] DEBUG[2276]: chan_sip.c:3689 __sip_xmit: Trying to put 'ACK sip:003' onto UDP socket destined for XXX.XXX.XXX.XXX:5060
-- SIP/ProveedorSIP-00000005 answered DAHDI/3-1
[Nov 8 18:55:48] DEBUG[1948]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - ProveedorSIP
[Nov 8 18:55:48] DEBUG[1948]: chan_sip.c:23236 sip_devicestate: Checking device state for peer ProveedorSIP
[Nov 8 18:55:48] DEBUG[1948]: devicestate.c:462 do_state_change: Changing state for SIP/ProveedorSIP - state 1 (Not in use)
[Nov 8 18:55:48] DEBUG[2868]: chan_dahdi.c:4768 dahdi_answer: Took DAHDI/3-1 off hook
[Nov 8 18:55:48] DEBUG[1948]: devicestate.c:442 devstate_event: device 'SIP/ProveedorSIP' state '1'
[Nov 8 18:55:48] DEBUG[2868]: chan_dahdi.c:7254 dahdi_indicate: Requested indication -1 on channel DAHDI/3-1
[Nov 8 18:55:48] DEBUG[1948]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for DAHDI - 3
[Nov 8 18:55:48] DEBUG[1948]: devicestate.c:462 do_state_change: Changing state for DAHDI/3 - state 2 (In use)
[Nov 8 18:55:48] DEBUG[1948]: devicestate.c:442 devstate_event: device 'DAHDI/3' state '2'
[Nov 8 18:55:48] DEBUG[2868]: features.c:2559 ast_bridge_call: bridge answer set, chan answer set
[Nov 8 18:55:48] DEBUG[2868]: chan_dahdi.c:7254 dahdi_indicate: Requested indication 20 on channel DAHDI/3-1
[Nov 8 18:55:48] DEBUG[2868]: rtp.c:2687 ast_rtp_new_source: Setting the marker bit due to a source update
[Nov 8 18:55:48] DEBUG[2184]: app_queue.c:1082 handle_statechange: Device 'SIP/ProveedorSIP' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Nov 8 18:55:48] DEBUG[2184]: app_queue.c:1082 handle_statechange: Device 'DAHDI/3' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Nov 8 18:55:51] DEBUG[2868]: chan_dahdi.c:6772 __dahdi_exception: Exception on 20, channel 3
[Nov 8 18:55:51] DEBUG[2868]: chan_dahdi.c:5806 dahdi_handle_event: Got event Event 27(27) on channel 3 (index 0)
[Nov 8 18:55:51] DEBUG[2868]: chan_dahdi.c:6672 dahdi_handle_event: Dunno what to do with event 27 on channel 3
[Nov 8 18:55:51] DEBUG[2868]: chan_dahdi.c:6772 __dahdi_exception: Exception on 20, channel 3
[Nov 8 18:55:51] DEBUG[2868]: chan_dahdi.c:5806 dahdi_handle_event: Got event Event 23(23) on channel 3 (index 0)
[Nov 8 18:55:51] DEBUG[2868]: chan_dahdi.c:6672 dahdi_handle_event: Dunno what to do with event 23 on channel 3
pulsar*CLI>
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:7463 sip_alloc: Allocating new SIP dialog for 565658f62f3e73eb...@127.0.0.1 - REGISTER (No RTP)
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:3367 registry_addref: SIP Registry sip.proveedorSIP.com: refcount now 3
[Nov 8 18:56:22] DEBUG[2276]: acl.c:506 ast_ouraddrfor: Found IP address for this socket
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:3810 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.2.127:5060
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:3367 registry_addref: SIP Registry sip.proveedorSIP.com: refcount now 4
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:11871 transmit_register: Scheduled a registration timeout for sip.proveedorSIP.com id #371
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:3402 initialize_initreq: Initializing initreq for method REGISTER - callid 565658f62f3e73eb...@127.0.0.1
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:11982 transmit_register: REGISTER attempt 1 to lo...@sip.proveedorSIP.com
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:3689 __sip_xmit: Trying to put 'REGISTER si' onto UDP socket destined for XXX.XXX.XXX.XXX:5060
[Nov 8 18:56:22] DEBUG[2276]: chan_sip.c:3359 registry_unref: SIP Registry sip.proveedorSIP.com: refcount now 3
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:4225 __sip_ack: Stopping retransmission on '565658f62f3e73eb...@127.0.0.1' of Request 168: Match Found
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:3400 initialize_initreq: Initializing already initialized SIP dialog 565658f62f3e73eb...@127.0.0.1 (presumably reinvite)
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:11982 transmit_register: REGISTER attempt 2 to lo...@sip.proveedorSIP.com
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:3689 __sip_xmit: Trying to put 'REGISTER si' onto UDP socket destined for XXX.XXX.XXX.XXX:5060
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:4225 __sip_ack: Stopping retransmission on '565658f62f3e73eb...@127.0.0.1' of Request 169: Match Found
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:18340 handle_response_register: Registration successful
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:18342 handle_response_register: Cancelling timeout 371
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:3359 registry_unref: SIP Registry sip.proveedorSIP.com: refcount now 2
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:3359 registry_unref: SIP Registry sip.proveedorSIP.com: refcount now 1
[Nov 8 18:56:23] DEBUG[2276]: chan_sip.c:3367 registry_addref: SIP Registry sip.proveedorSIP.com: refcount now 2
[Nov 8 18:56:25] DEBUG[2276]: chan_sip.c:7463 sip_alloc: Allocating new SIP dialog for 1698e0605f5e4753...@127.0.0.1 - OPTIONS (No RTP)
[Nov 8 18:56:25] DEBUG[2276]: acl.c:506 ast_ouraddrfor: Found IP address for this socket
[Nov 8 18:56:25] DEBUG[2276]: chan_sip.c:3810 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.2.127:5060
[Nov 8 18:56:25] DEBUG[2276]: chan_sip.c:3402 initialize_initreq: Initializing initreq for method OPTIONS - callid 7c931d3367a0fb17...@192.168.2.127
[Nov 8 18:56:25] DEBUG[2276]: chan_sip.c:3689 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for XXX.XXX.XXX.XXX:5060
[Nov 8 18:56:25] DEBUG[2276]: chan_sip.c:4225 __sip_ack: Stopping retransmission on '7c931d3367a0fb17...@192.168.2.127' of Request 102: Match Found
[Nov 8 18:56:25] DEBUG[2276]: chan_sip.c:5930 sip_destroy: Destroying SIP dialog 7c931d3367a0fb17...@192.168.2.127
[Nov 8 18:56:29] DEBUG[2868]: chan_dahdi.c:6772 __dahdi_exception: Exception on 20, channel 3
[Nov 8 18:56:29] DEBUG[2868]: chan_dahdi.c:5806 dahdi_handle_event: Got event On hook(1) on channel 3 (index 0)
[Nov 8 18:56:29] DEBUG[2868]: chan_dahdi.c:2723 dahdi_disable_ec: Disabled echo cancellation on channel 3
[Nov 8 18:56:29] DEBUG[2868]: channel.c:5044 ast_generic_bridge: Didn't get a frame from channel: DAHDI/3-1
[Nov 8 18:56:29] DEBUG[2868]: rtp.c:2687 ast_rtp_new_source: Setting the marker bit due to a source update
[Nov 8 18:56:29] DEBUG[2868]: channel.c:5472 ast_channel_bridge: Bridge stops bridging channels DAHDI/3-1 and SIP/proveedorSIP-00000005
[Nov 8 18:56:29] DEBUG[2868]: pbx.c:3520 pbx_substitute_variables_helper_full: Function result is 'yes'
[Nov 8 18:56:29] DEBUG[2868]: pbx.c:3692 pbx_extension_helper: Launching 'NoOp'
-- Executing [h@outboundfax:1] NoOp("DAHDI/3-1", "FAXOPT(ecm) : yes") in new stack
[Nov 8 18:56:29] DEBUG[2868]: cdr_addon_mysql.c:307 mysql_log: Inserting a CDR record.
[Nov 8 18:56:29] DEBUG[2868]: cdr_addon_mysql.c:310 mysql_log: SQL command as follows: INSERT INTO cdr (calldate,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid) VALUES ('2012-11-08 18:55:31','4003','9XXXXXXXX','outboundfax','DAHDI/3-1','SIP/proveedorSIP-00000005','Dial','SIP/proveedorSIP/00349XXXXXXXX,20','58','41','ANSWERED','3','1352397331.18')
[Nov 8 18:56:29] DEBUG[2868]: cdr_radius.c:207 radius_log: Unable to create RADIUS record. CDR not recorded!
[Nov 8 18:56:29] DEBUG[2868]: res_config_sqlite.c:829 cdr_handler: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"device" <4003>','4003','9XXXXXXXX','outboundfax','DAHDI/3-1','SIP/proveedorSIP-00000005','Dial','SIP/proveedorSIP/00349XXXXXXXX,20','2012-11-08 18:55:31','2012-11-08 18:55:48','2012-11-08 18:56:29','58','41','ANSWERED','DOCUMENTATION','1352397331.18')
[Nov 8 18:56:29] DEBUG[2868]: channel.c:1820 ast_hangup: Hanging up channel 'SIP/proveedorSIP-00000005'
[Nov 8 18:56:29] DEBUG[2868]: chan_sip.c:6147 sip_hangup: Hangup call SIP/proveedorSIP-00000005, SIP callid 7b8b2ca875e826c4...@192.168.2.127
[Nov 8 18:56:29] DEBUG[2868]: chan_sip.c:3689 __sip_xmit: Trying to put 'BYE sip:003' onto UDP socket destined for XXX.XXX.XXX.XXX:5060
[Nov 8 18:56:29] DEBUG[2868]: app_dial.c:2317 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Nov 8 18:56:29] DEBUG[2868]: pbx.c:4317 __ast_pbx_run: Spawn extension (outboundfax,9XXXXXXXX,3) exited non-zero on 'DAHDI/3-1'
== Spawn extension (outboundfax, 9XXXXXXXX, 3) exited non-zero on 'DAHDI/3-1'
[Nov 8 18:56:29] DEBUG[2868]: channel.c:1715 ast_softhangup_nolock: Soft-Hanging up channel 'DAHDI/3-1'
[Nov 8 18:56:29] DEBUG[2868]: channel.c:1820 ast_hangup: Hanging up channel 'DAHDI/3-1'
[Nov 8 18:56:29] DEBUG[2868]: chan_dahdi.c:4325 dahdi_hangup: dahdi_hangup(DAHDI/3-1)
[Nov 8 18:56:29] DEBUG[2868]: chan_dahdi.c:4370 dahdi_hangup: Hangup: channel: 3 index = 0, normal = 20, callwait = -1, thirdcall = -1
[Nov 8 18:56:29] DEBUG[2868]: chan_dahdi.c:4905 dahdi_setoption: Set option TDD MODE, value: OFF(0) on DAHDI/3-1
[Nov 8 18:56:29] DEBUG[2868]: chan_dahdi.c:2658 update_conf: Updated conferencing on 3, with 0 conference users
[Nov 8 18:56:29] DEBUG[1948]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - proveedorSIP
[Nov 8 18:56:29] DEBUG[1948]: chan_sip.c:23236 sip_devicestate: Checking device state for peer proveedorSIP
-- Hungup 'DAHDI/3-1'
[Nov 8 18:56:29] DEBUG[1948]: devicestate.c:462 do_state_change: Changing state for SIP/proveedorSIP - state 1 (Not in use)
[Nov 8 18:56:29] DEBUG[1948]: devicestate.c:442 devstate_event: device 'SIP/proveedorSIP' state '1'
[Nov 8 18:56:29] DEBUG[1948]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for DAHDI - 3
[Nov 8 18:56:29] DEBUG[1948]: devicestate.c:462 do_state_change: Changing state for DAHDI/3 - state 0 (Unknown)
[Nov 8 18:56:29] DEBUG[1948]: devicestate.c:442 devstate_event: device 'DAHDI/3' state '0'
[Nov 8 18:56:29] DEBUG[2184]: app_queue.c:1082 handle_statechange: Device 'SIP/proveedorSIP' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Nov 8 18:56:29] DEBUG[2184]: app_queue.c:1082 handle_statechange: Device 'DAHDI/3' changed to state '0' (Unknown) but we don't care because they're not a member of any queue.
[Nov 8 18:56:29] DEBUG[2276]: chan_sip.c:4225 __sip_ack: Stopping retransmission on '7b8b2ca875e826c4...@192.168.2.127' of Request 104: Match Found
[Nov 8 18:56:29] DEBUG[2276]: chan_sip.c:5930 sip_destroy: Destroying SIP dialog 7b8b2ca875e826c4...@192.168.2.127
pulsar*CLI>
pulsar*CLI>
pulsar*CLI>
pulsar*CLI>
pulsar*CLI>
pulsar*CLI>
pulsar*CLI>
pulsar*CLI>
pulsar*CLI>
pulsar*CLI>
pulsar*CLI> core set debug 0
Core debug is now OFF
pulsar*CLI>
Mi extension.conf tengo lo siguiente en el contexto de salida de FAX :
[outboundfax]
exten => _9.,1,NoOp()
exten => _9.,n,Set(FAXOPT(gateway)=yes)
exten => _9.,n,Dial(SIP/freevoipdeal/0034${EXTEN},20)
exten => _9.,n,Set(FAXOPT(ecm)=yes)
exten => _9.,n,Set(FAXOPT(headerinfo)=Fax from MYCOMPANY +34 XXXXXXXXX)
exten => _9.,n,Set(FAXOPT(localstationid)=XXXXXXXXX)
exten => _9.,n,Set(FAXOPT(maxrate)=14400)
exten => _9.,n,Set(FAXOPT(minrate)=2400)exten => _9.,n,SendFAX(/tmp/${FAXFILE},d)exten => h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
segunda duda : SendFAX, no se que fichero decirle que coga, el fax envia los datos y asterisk lo tiene que coger, los almacena en algun sitio?¿?
Como veis, voy algo perdido, cualquier aclaración será de agradecer.
Saludos!
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