Re: [Asterisk-Users] fax pass-through
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
[EMAIL PROTECTED] - 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
[EMAIL PROTECTED]
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
'[EMAIL PROTECTED]' 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:
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 (Ri
[Asterisk-Users] fax pass-through
hi,
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
log:
Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for
[EMAIL PROTECTED] - 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
[EMAIL PROTECTED]
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
'[EMAIL PROTECTED]' 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:
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]
[Asterisk-Users] FAX pass through
I would like to receive/send faxes, from a SIP gateway. Asterisk can do that ???. A phone can receive the fax call, then transfer to the SIP gateway, where a FAX machine is connected. We have problem, but I don t know is the problem is the gateway (Addpac)??? Thanks Alex Ternero ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax pass through on te410p
I had the same problem at one site. We could not receive faxes with spandsp reliably. Our solution that seems to have worked with no problems so far was to use a SPA-2000 to a fax machine. - Original Message - From: "Kevin Brennan" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, April 09, 2005 7:16 AM Subject: Re: [Asterisk-Users] fax pass through on te410p Ok - point taken - but we're running Asterisk as a SIP/PSTN gateway and we don't seem to have any other noticable problems, ok fax is more sensitive. We've tried different versions of spandsp and it does not fix anything, ok perhaps this shows problem is not spandsp - so where/how to start looking for a fix - any pointers anyone. If your hardware isn't getting clean data to spandsp, why should it be able to get clean data to a hylafax box? Unless you fix the config problem that stops spandsp working, there is no reason to expect a pass-through to a modem bank and hylafax to work. Regards, Steve Kevin Brennan wrote: > We are using spandsp but find it unusable in a commercial environment, > we are looking at changing to a dedicated hylafax server using an > eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a > reliable config our only uncertainty is how good Asterisk is at > handling pass through fax on a te410p. Has anybody got good/bad > experience with similar setup ? > > Br/Kevin Brennan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax pass through on te410p
Ok - point taken - but we're running Asterisk as a SIP/PSTN gateway and we don't seem to have any other noticable problems, ok fax is more sensitive. We've tried different versions of spandsp and it does not fix anything, ok perhaps this shows problem is not spandsp - so where/how to start looking for a fix - any pointers anyone. > If your hardware isn't getting clean data to spandsp, why should it be > able to get clean data to a hylafax box? Unless you fix the config > problem that stops spandsp working, there is no reason to expect a > pass-through to a modem bank and hylafax to work. > > Regards, > Steve > > > Kevin Brennan wrote: > > > We are using spandsp but find it unusable in a commercial environment, > > we are looking at changing to a dedicated hylafax server using an > > eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a > > reliable config our only uncertainty is how good Asterisk is at > > handling pass through fax on a te410p. Has anybody got good/bad > > experience with similar setup ? > > > > Br/Kevin Brennan > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax pass through on te410p
If your hardware isn't getting clean data to spandsp, why should it be able to get clean data to a hylafax box? Unless you fix the config problem that stops spandsp working, there is no reason to expect a pass-through to a modem bank and hylafax to work. Regards, Steve Kevin Brennan wrote: We are using spandsp but find it unusable in a commercial environment, we are looking at changing to a dedicated hylafax server using an eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a reliable config our only uncertainty is how good Asterisk is at handling pass through fax on a te410p. Has anybody got good/bad experience with similar setup ? Br/Kevin Brennan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax pass through on te410p
We are using spandsp but find it unusable in a commercial environment, we are looking at changing to a dedicated hylafax server using an eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a reliable config our only uncertainty is how good Asterisk is at handling pass through fax on a te410p. Has anybody got good/bad experience with similar setup ? Br/Kevin Brennan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
