[HELP] PC with Fax modem - PAP2T - Asterisk@home

I sent fax with my PC via my orignal landlines.

I recently installed Asterisk@home, and I got PAP2T ATA for my PC to get extension number. But I can’t send fax with my PC now. I can hear both side handshaking. Please help. (I use VoipJet for outgoing trunk).

Can anyone help? thanks in advance.

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Aug 22 10:12:14 DEBUG[2600] chan_sip.c: Setting NAT on RTP to 0
Aug 22 10:12:14 DEBUG[2600] chan_sip.c: Checking SIP call limits for device 260
Aug 22 10:12:14 DEBUG[2600] chan_sip.c: build_route: Contact hop: 260
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing Macro(“SIP/260-6aa5”, “dialout-trunk|2|15555555555|”) in new stack
Aug 22 10:12:14 DEBUG[26110] pbx.c: Expression result is '1’
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing GotoIf(“SIP/260-6aa5”, “1?3:2)”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Goto (macro-dialout-trunk,s,3)
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing Macro(“SIP/260-6aa5”, “user-callerid”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing DBget(“SIP/260-6aa5”, “AMPUSER=DEVICE/260/user”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – DBget: varname=AMPUSER, family=DEVICE, key=260/user
Aug 22 10:12:14 VERBOSE[26110] logger.c: – DBget: set variable AMPUSER to 260
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing DBget(“SIP/260-6aa5”, “AMPUSERCIDNAME=AMPUSER/260/cidname”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=260/cidname
Aug 22 10:12:14 VERBOSE[26110] logger.c: – DBget: set variable AMPUSERCIDNAME to rem
Aug 22 10:12:14 DEBUG[26110] pbx.c: Expression result is '0’
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing GotoIf(“SIP/260-6aa5”, “0?5”) in new stack
Aug 22 10:12:14 DEBUG[26110] pbx.c: Not taking any branch
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing SetCallerID(“SIP/260-6aa5”, ““rem” <260>”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing NoOp(“SIP/260-6aa5”, “Using CallerID “rem” <260>”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing Macro(“SIP/260-6aa5”, “record-enable|260|OUT”) in new stack
Aug 22 10:12:14 DEBUG[26110] pbx.c: Function result is '0’
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing GotoIf(“SIP/260-6aa5”, “0 > 0?2:4”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Goto (macro-record-enable,s,4)
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing AGI(“SIP/260-6aa5”, “recordingcheck|20060822-101214|1156255934.64”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
Aug 22 10:12:14 VERBOSE[26110] logger.c: recordingcheck|20060822-101214|1156255934.64: Outbound recording not enabled
Aug 22 10:12:14 VERBOSE[26110] logger.c: – AGI Script recordingcheck completed, returning 0
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing NoOp(“SIP/260-6aa5”, “No recording needed”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing Macro(“SIP/260-6aa5”, “outbound-callerid|2”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing DBget(“SIP/260-6aa5”, “USEROUTCID=AMPUSER/260/outboundcid”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – DBget: varname=USEROUTCID, family=AMPUSER, key=260/outboundcid
Aug 22 10:12:14 VERBOSE[26110] logger.c: – DBget: set variable USEROUTCID to
Aug 22 10:12:14 DEBUG[26110] pbx.c: Expression result is '0’
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing GotoIf(“SIP/260-6aa5”, “0?4”) in new stack
Aug 22 10:12:14 DEBUG[26110] pbx.c: Not taking any branch
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing SetCallerID(“SIP/260-6aa5”, “555-555-5555”) in new stack
Aug 22 10:12:14 DEBUG[26110] pbx.c: Expression result is '1’
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing GotoIf(“SIP/260-6aa5”, “1?6”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Goto (macro-outbound-callerid,s,6)
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing NoOp(“SIP/260-6aa5”, “CallerID set to 5555550505”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing SetGroup(“SIP/260-6aa5”, “OUT_2”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing CheckGroup(“SIP/260-6aa5”, “5”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing SetVar(“SIP/260-6aa5”, “DIAL_NUMBER=15555555555”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing SetVar(“SIP/260-6aa5”, “DIAL_TRUNK=2”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Executing AGI(“SIP/260-6aa5”, “fixlocalprefix”) in new stack
Aug 22 10:12:14 VERBOSE[26110] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
Aug 22 10:12:15 VERBOSE[26110] logger.c: – AGI Script fixlocalprefix completed, returning 0
Aug 22 10:12:15 VERBOSE[26110] logger.c: – Executing SetVar(“SIP/260-6aa5”, “OUTNUM=15555555555”) in new stack
Aug 22 10:12:15 VERBOSE[26110] logger.c: – Executing Cut(“SIP/260-6aa5”, “custom=OUT_2|:|1”) in new stack
Aug 22 10:12:15 WARNING[26110] ast_expr2.y: non-numeric argument
Aug 22 10:12:15 DEBUG[26110] pbx.c: Expression result is '0’
Aug 22 10:12:15 VERBOSE[26110] logger.c: – Executing GotoIf(“SIP/260-6aa5”, “0?16”) in new stack
Aug 22 10:12:15 DEBUG[26110] pbx.c: Not taking any branch
Aug 22 10:12:15 VERBOSE[26110] logger.c: – Executing Dial(“SIP/260-6aa5”, “IAX2/voipjet/15555555555”) in new stack
Aug 22 10:12:15 VERBOSE[26110] logger.c: – Called voipjet/15555555555
Aug 22 10:12:15 VERBOSE[2597] logger.c: – Call accepted by 64.34.45.100 (format ulaw)
Aug 22 10:12:15 VERBOSE[2597] logger.c: – Format for call is ulaw
Aug 22 10:12:17 VERBOSE[26110] logger.c: – IAX2/voipjet-1 is ringing
Aug 22 10:12:17 VERBOSE[26110] logger.c: – IAX2/voipjet-1 is making progress passing it to SIP/260-6aa5
Aug 22 10:12:17 DEBUG[2597] chan_iax2.c: Ooh, voice format changed to 4
Aug 22 10:12:17 NOTICE[26110] rtp.c: Unknown RTP codec 100 received
Aug 22 10:12:25 DEBUG[2600] chan_sip.c: Auto destroying call '5ff53f2-480b9fb5@192.168.1.103’
Aug 22 10:12:32 VERBOSE[26110] logger.c: – IAX2/voipjet-1 answered SIP/260-6aa5
Aug 22 10:12:32 DEBUG[2600] chan_sip.c: Stopping retransmission on ‘d1a3c463-8257cdf5@192.168.1.103’ of Response 102: Match Found
Aug 22 10:13:04 DEBUG[26110] channel.c: Didn’t get a frame from channel: SIP/260-6aa5
Aug 22 10:13:04 DEBUG[26110] channel.c: Bridge stops bridging channels SIP/260-6aa5 and IAX2/voipjet-1
Aug 22 10:13:04 DEBUG[26110] chan_iax2.c: We’re hanging up IAX2/voipjet-1 now…
Aug 22 10:13:04 VERBOSE[26110] logger.c: – Hungup 'IAX2/voipjet-1’
Aug 22 10:13:04 DEBUG[26110] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Aug 22 10:13:04 VERBOSE[26110] logger.c: == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on ‘SIP/260-6aa5’ in macro 'dialout-trunk’
Aug 22 10:13:04 VERBOSE[26110] logger.c: == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/260-6aa5’
Aug 22 10:13:04 DEBUG[26110] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Aug 22 10:13:04 DEBUG[26110] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr

(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2006-08-22

10:12:14’,‘5555550505’,‘5555550505’,‘15555555555’,‘from-internal’, ‘SIP/260-6aa5’,‘IAX2/voipjet-1’,‘Dial’,‘IAX2/voipjet/15555555555’,50,32,‘ANSWERED’,3,’’,‘1156255934.64’)

Have you tried hylafax.
hylafax.org/

Maaan im too lazy to reall all that, are you using g729 or 711?

are you using g729 or 711?
G711

Only PBX is in Linux, other PCs are with Windows. The PC with Fax modem and other fax applications is also Windows based.

What verified:
PAP2 is configured G711 only.

What ese I need to do? thanks in advance.

Can anyone explain:

Aug 22 10:12:17 DEBUG[2597] chan_iax2.c: Ooh, voice format changed to 4 Aug 22 10:12:17 NOTICE[26110] rtp.c: Unknown RTP codec 100 received
Aug 22 10:12:25 DEBUG[2600] chan_sip.c: Auto destroying call ‘5ff53f2-480b9fb5@192.168.1.103’

Thanks.

make sure t.38 is turned off

[quote] make sure t.38 is turned off
[/quote]

How do I do that? thanks

you might also check the PAP2 config, ensure all echo cancellation is off.

on our linksys SPA-2002’s, the option is under the advanced admin page, and can be configured on a line-by-line basis. if you REALLY wanna get into it, read here:

voip-info.org/wiki-Asterisk+fax