[Asterisk-Users] fax pass-through

marek cervenka cervajs at fpf.slu.cz
Tue Feb 14 14:52:23 MST 2006 

> after upgrade from 1.0.x to 1.2.x i cannot send faxes
> my topology:
> PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 
> fax

is there someone with this scenario? it is working?
thanks

(ip connectivity is good, codec alaw, 0% success)

> log:
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 
> 20d700003cb20000 at 192.168.1.209 - INVITE (With RTP)
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command 
> in SIP INVITE
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: * SIP extension value: 1 for call 
> 20d700003cb20000 at 192.168.1.209
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Setting NAT on RTP to 524288
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received ACK (6) - Command in 
> SIP ACK
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Stopping retransmission on 
> '20d700003cb20000 at 192.168.1.209' of Response 3727: Match Found
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command 
> in SIP INVITE
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Setting NAT on RTP to 524288
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Checking SIP call limits for device 
> 46
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Updating call counter for incoming 
> call
> Feb 13 23:50:35 DEBUG[27914] chan_sip.c: build_route: Contact hop: 
> <sip:46 at 192.168.1.209>
> Feb 13 23:50:35 DEBUG[27904] chan_sip.c: Checking device state for peer 46
> Feb 13 23:50:35 DEBUG[27904] devicestate.c: Changing state for SIP/46 - state 
> 2 (In use)
> Feb 13 23:50:35 DEBUG[28047] pbx.c: Launching 'Goto'
> Feb 13 23:50:35 DEBUG[28048] app_queue.c: Device 'SIP/46' changed to state 
> '2' (In use)
> Feb 13 23:50:35 VERBOSE[28047] logger.c:     -- Executing Goto("SIP/46-62bb", 
> "pstn|54|1") in new stack
> Feb 13 23:50:35 VERBOSE[28047] logger.c:     -- Goto (pstn,54,1)
> Feb 13 23:50:35 DEBUG[28047] chan_iax2.c: peer: 192.168.9.35, username: voip, 
> password: test, context: (null)
> Feb 13 23:50:35 VERBOSE[27913] logger.c:     -- Call accepted by 192.168.9.35 
> (format g729)
> Feb 13 23:50:35 DEBUG[28047] pbx.c: Launching 'Macro'
> Feb 13 23:50:35 VERBOSE[28047] logger.c:     -- Executing 
> Macro("SIP/46-62bb", "stdial|Zap/g1/54|300|tT") in new stack
> Feb 13 23:50:35 DEBUG[28047] pbx.c: Launching 'NoOp'
> Feb 13 23:50:35 VERBOSE[28047] logger.c:     -- Executing NoOp("SIP/46-62bb", 
> "46") in new stack
> Feb 13 23:50:35 DEBUG[28047] pbx.c: Launching 'Dial'
> Feb 13 23:50:35 VERBOSE[28047] logger.c:     -- Executing Dial("SIP/46-62bb", 
> "Zap/g1/54||tT") in new stack
> Feb 13 23:50:35 DEBUG[28047] chan_zap.c: Using channel 1
> Feb 13 23:50:35 DEBUG[27904] devicestate.c: Changing state for Zap/1 - state 
> 2 (In use)
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable 
> STACK-macro-stdial-s-2.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable MACRO_DEPTH.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable 
> STACK-macro-stdial-s-1.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable ARG3.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable ARG2.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable ARG1.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable MACRO_PRIORITY.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable MACRO_CONTEXT.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable MACRO_EXTEN.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable STACK-pstn-54-1.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable 
> STACK-from_customers-54-1.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable SIPCALLID.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable SIPUSERAGENT.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable SIPDOMAIN.
> Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable SIPURI.
> Feb 13 23:50:35 DEBUG[28049] app_queue.c: Device 'Zap/1' changed to state '2' 
> (In use)
> Feb 13 23:50:35 VERBOSE[28047] logger.c:     -- Requested transfer 
> capability: 0x00 - SPEECH
> Feb 13 23:50:35 VERBOSE[28047] logger.c:     -- Called g1/54
> Feb 13 23:50:35 DEBUG[28047] channel.c: Set channel Zap/1-1 to read format 
> alaw
> Feb 13 23:50:35 DEBUG[28047] channel.c: Set channel SIP/46-62bb to write 
> format alaw
> Feb 13 23:50:35 DEBUG[28047] channel.c: Set channel SIP/46-62bb to read 
> format alaw
> Feb 13 23:50:35 DEBUG[28047] channel.c: Set channel Zap/1-1 to write format 
> alaw
> Feb 13 23:50:35 DEBUG[27904] devicestate.c: Changing state for Zap/1 - state 
> 2 (In use)
> Feb 13 23:50:35 DEBUG[28050] app_queue.c: Device 'Zap/1' changed to state '2' 
> (In use)
> Feb 13 23:50:35 DEBUG[28047] rtp.c: Ooh, format changed from unknown to alaw
> Feb 13 23:50:35 DEBUG[27908] chan_zap.c: Queuing frame from 
> PRI_EVENT_PROCEEDING on channel 0/1 span 1
> Feb 13 23:50:35 VERBOSE[28047] logger.c:     -- Zap/1-1 is proceeding passing 
> it to SIP/46-62bb
> Feb 13 23:50:35 DEBUG[28047] rtp.c: RTP NAT: Got audio from other end. Now 
> sending to address 213.155.226.151:5004
> Feb 13 23:50:36 DEBUG[27908] chan_zap.c: Enabled echo cancellation on channel 
> 1
> Feb 13 23:50:36 DEBUG[27904] devicestate.c: Changing state for Zap/1 - state 
> 6 (Ringing)
> Feb 13 23:50:36 VERBOSE[28047] logger.c:     -- Zap/1-1 is ringing
> Feb 13 23:50:36 DEBUG[28051] app_queue.c: Device 'Zap/1' changed to state '6' 
> (Ringing)
> Feb 13 23:50:54 DEBUG[27908] chan_zap.c: Echo cancellation already on
> Feb 13 23:50:54 DEBUG[27904] channel.c: Avoiding initial deadlock for 
> 'Zap/1-1'
> Feb 13 23:50:54 VERBOSE[28047] logger.c: << [ TYPE: Control (4) SUBCLASS: 
> Answer (4) ] [Zap/1-1]
> Feb 13 23:50:54 VERBOSE[28047] logger.c:     -- Zap/1-1 answered SIP/46-62bb
> Feb 13 23:50:54 DEBUG[28047] channel.c: Set channel SIP/46-62bb to read 
> format alaw
> Feb 13 23:50:54 DEBUG[28047] channel.c: Set channel Zap/1-1 to write format 
> alaw
> Feb 13 23:50:54 DEBUG[28047] channel.c: Set channel Zap/1-1 to read format 
> alaw
> Feb 13 23:50:54 DEBUG[28047] channel.c: Set channel SIP/46-62bb to write 
> format alaw
> Feb 13 23:50:54 DEBUG[28047] chan_sip.c: sip_answer(SIP/46-62bb)
> Feb 13 23:50:54 DEBUG[27904] devicestate.c: Changing state for Zap/1 - state 
> 2 (In use)
> Feb 13 23:50:54 DEBUG[27904] chan_sip.c: Checking device state for peer 46
> Feb 13 23:50:54 DEBUG[27904] devicestate.c: Changing state for SIP/46 - state 
> 2 (In use)
> Feb 13 23:50:54 DEBUG[28052] app_queue.c: Device 'Zap/1' changed to state '2' 
> (In use)
> Feb 13 23:50:54 DEBUG[28053] app_queue.c: Device 'SIP/46' changed to state 
> '2' (In use)
> Feb 13 23:50:54 DEBUG[27914] chan_sip.c: **** Received ACK (6) - Command in 
> SIP ACK
> Feb 13 23:50:54 DEBUG[27914] chan_sip.c: Stopping retransmission on 
> '20d700003cb20000 at 192.168.1.209' of Response 3728: Match Found
> Feb 13 23:50:56 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for (No 
> Call-ID) - OPTIONS (No RTP)
> Feb 13 23:50:56 DEBUG[27914] chan_sip.c: Stopping retransmission on 
> '19b7d4d226e0e1134b97362158429920 at 212.71.129.36' of Request 102: Match Found
> Feb 13 23:50:56 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 
> 19b7d4d226e0e1134b97362158429920 at 212.71.129.36 - SIP/2.0 (No RTP)
> Feb 13 23:50:56 DEBUG[27914] chan_sip.c: That's odd...  Got a response on a 
> call we dont know about. Cseq 102 Cmd SIP/2.0
> Feb 13 23:50:58 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command 
> in SIP INVITE
> Feb 13 23:50:58 VERBOSE[27914] logger.c:     -- Music class default requested 
> but no musiconhold loaded.
> Feb 13 23:50:58 DEBUG[27914] chan_sip.c: **** Received ACK (6) - Command in 
> SIP ACK
> Feb 13 23:50:58 DEBUG[27914] chan_sip.c: Stopping retransmission on 
> '20d700003cb20000 at 192.168.1.209' of Response 3729: Match Found
> Feb 13 23:51:53 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command 
> in SIP INVITE
> Feb 13 23:51:53 DEBUG[28047] rtp.c: RTP NAT: Got audio from other end. Now 
> sending to address 213.155.226.151:5004
> Feb 13 23:51:53 DEBUG[28047] rtp.c: Difference is 436152, ms is 54539
> Feb 13 23:51:53 DEBUG[27914] chan_sip.c: **** Received ACK (6) - Command in 
> SIP ACK
> Feb 13 23:51:53 DEBUG[27914] chan_sip.c: Stopping retransmission on 
> '20d700003cb20000 at 192.168.1.209' of Response 3730: Match Found
> Feb 13 23:51:56 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for (No 
> Call-ID) - OPTIONS (No RTP)
> Feb 13 23:51:56 DEBUG[27914] chan_sip.c: Stopping retransmission on 
> '5615f3cf27e0b9c301b6d88d706ea82d at 212.71.129.36' of Request 102: Match Found
> Feb 13 23:51:56 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 
> 5615f3cf27e0b9c301b6d88d706ea82d at 212.71.129.36 - SIP/2.0 (No RTP)
> Feb 13 23:51:56 DEBUG[27914] chan_sip.c: That's odd...  Got a response on a 
> call we dont know about. Cseq 102 Cmd SIP/2.0
> Feb 13 23:52:00 DEBUG[27914] chan_sip.c: **** Received BYE (8) - Command in 
> SIP BYE
> Feb 13 23:52:00 DEBUG[28047] channel.c: Didn't get a frame from channel: 
> SIP/46-62bb
> Feb 13 23:52:00 DEBUG[28047] channel.c: Bridge stops bridging channels 
> SIP/46-62bb and Zap/1-1
> Feb 13 23:52:00 DEBUG[28047] channel.c: Hanging up channel 'Zap/1-1'
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: zt_hangup(Zap/1-1)
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: Set option AUDIO MODE, value: ON(1) 
> on Zap/1-1
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: Hangup: channel: 1 index = 0, normal 
> = 10, callwait = -1, thirdcall = -1
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: Not yet hungup...  Calling hangup 
> once with icause, and clearing call
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: disabled echo cancellation on 
> channel 1
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: Set option TDD MODE, value: OFF(0) 
> on Zap/1-1
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: Updated conferencing on 1, with 0 
> conference users
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: Set option AUDIO MODE, value: OFF(0) 
> on Zap/1-1
> Feb 13 23:52:00 DEBUG[28047] chan_zap.c: disabled echo cancellation on 
> channel 1
> Feb 13 23:52:00 VERBOSE[28047] logger.c:     -- Hungup 'Zap/1-1'
> Feb 13 23:52:00 DEBUG[28047] app_dial.c: Exiting with DIALSTATUS=ANSWER.
> Feb 13 23:52:00 DEBUG[28047] app_macro.c: Spawn extension (macro-stdial,s,2) 
> exited non-zero on 'SIP/46-62bb' in macro 'stdial'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Spawn extension (macro-stdial,s,2) exited 
> non-zero on 'SIP/46-62bb'
> Feb 13 23:52:00 DEBUG[27904] devicestate.c: Changing state for Zap/1 - state 
> 0 (Unknown)
> Feb 13 23:52:00 DEBUG[28054] app_queue.c: Device 'Zap/1' changed to state '0' 
> (Unknown)
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '46'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '46'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '54'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'pstn'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'SIP/46-62bb'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Zap/1-1'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Dial'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'Zap/g1/54||tT'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:50:35'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:50:54'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '2006-02-13 23:52:00'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '85'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '66'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'ANSWERED'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is 'DOCUMENTATION'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '(null)'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '1139871035.6'
> Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '(null)'
>
>
> any ideas?
>
> ---------------------------------------
> Marek Cervenka
> =======================================
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>

---------------------------------------
Marek Cervenka
=======================================

More information about the asterisk-users mailing list