[asterisk-users] TDM400P with FXS module problem
Hi list I conncte my Panasonic KX-TCD705TW cordless phone to TDM400P FXS module. When the system boots or reboots, the LED on the backlit of TDM400P OFTEN gets off and dmesg shows problem with FXS module, as follows Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.0 Echo Canceller: MG2 ACPI: PCI interrupt :02:08.0[A] - GSI 19 (level, low) - IRQ 177 Freshmaker version: 71 Freshmaker passed register test Timeout waiting for calibration of module 0 Timeout waiting for calibration of module 0 Proslic Failed on Second Attempt to Auto Calibrate Proslic Failed on Second Attempt to Calibrate Manually. (Try -DNO_CALIBRATION in Makefile) Module 0: FAILED FXS (FCC) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules) Registered tone zone 14 (Taiwan) When this occurs, no dialtone is heard after off-hook. I have to rmmod wctdm, unplug the phone cable from TDM400P, modprobe wctdm, and plug phone cable to make this line work. The point is ztcfg without rmmod wctdm cannot make working. I run asterisk 1.4.0 / zaptel 1.4.0 on Debian 3.1r4. I compile from source. Any help or recommendation will be appreciated. ___ --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
Re: [asterisk-users] background() with m option
I have same problem and no mailing list response. I suggest we go for
reporting bug.
- Original Message -
From: Jack Wei [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, January 26, 2007 1:16 AM
Subject: [asterisk-users] background() with m option
Hi...
In my dialplan, I have the following:
exten = s,1,Background(${RECORDING}|m)
exten = s,n,Voicemail(${DID_NO})
exten = 0,1,Voicemail(${DID_NO})
exten = a,1,VoiceMailMain(${DID_NO})
exten = h,1,Hangup
In version 1.2, when I hit 0 during the playback, I will be directed to
voicemail. But in verison 1.4, the call hangs up.
[Jan 24 16:05:37] DTMF[5754]: channel.c:2148 __ast_read: DTMF begin '0'
received on SIP/5060-08c53e68
[Jan 24 16:05:37] DTMF[5754]: channel.c:2128 __ast_read: DTMF end '0'
received on SIP/5060-08c53e68
== Spawn extension (play_recording, s, 1) exited non-zero on
'SIP/5060-08c53e68'
-- Executing [EMAIL PROTECTED]:1] Hangup(SIP/5060-08c53e68, ) in new
stack
== Spawn extension (play_recording, h, 1) exited non-zero on
'SIP/5060-08c53e68'
Does anyone tell me why this is happening?
Thanks,
Jack
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[asterisk-users] cmd Backgound problem with option m
Hi list
I encountered problem in using Background command. Below is my
extensions.conf.
[mainmenu]
exten = 4,1,Wait(1)
exten = 4,2,Background(thank-you-for-calling)
exten = 4,3,Goto(n01|s|1)
[n01]
exten = s,1,NoOp(${CONTEXT})
exten = s,2,Background(thank-you-cooperation|m)
exten = s,3,WaitExten()
exten = s,4,Playback(digits/pound)
exten = 1,1,Playback(digits/1)
exten = i,1,Playback(digits/star)
Without m option, everything's fine.
If m option is present and when sound is playing,
- pressing 1 terminates the call and does not goto ext 1
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-for-calling)
in new stack
-- Playing 'thank-you-for-calling' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1,
thank-you-cooperation|m) in
new stack
-- Playing 'thank-you-cooperation' (language 'en')
== Spawn extension (n01, s, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
I also tried Background(thank-you-cooperation|m||n01). The result is
- pressing 1 goto ext i
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation)
in new stack
-- Playing 'thank-you-cooperation' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1,
thank-you-cooperation|m||n01)
in new stack
-- Playing 'thank-you-cooperation' (language '')
-- Sent into invalid extension 'E8' in context 'n01' on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Playback(Zap/1-1, digits/star) in new
stack
-- Playing 'digits/star' (language 'en')
== Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'
NOTICE that * tries to go to ext 'E8' which is a French alphabet e with
grave accent.
DTMF detection problem? but if context option and m option of Background is
not specified, everything works well.
any help will be appreciated.
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[asterisk-users] cmd Backgound problem with option m
Hi list
I encountered problem in using Background command. Below is my
extensions.conf.
[mainmenu]
exten = 4,1,Wait(1)
exten = 4,2,Background(thank-you-for-calling)
exten = 4,3,Goto(n01|s|1)
[n01]
exten = s,1,NoOp(${CONTEXT})
exten = s,2,Background(thank-you-cooperation|m)
exten = s,3,WaitExten()
exten = s,4,Playback(digits/pound)
exten = 1,1,Playback(digits/1)
exten = i,1,Playback(digits/star)
Without m option, everything's fine.
If m option is present and when sound is playing,
- pressing 1 terminates the call and does not goto ext 1
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-for-calling)
in new stack
-- Playing 'thank-you-for-calling' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation|m) in
new stack
-- Playing 'thank-you-cooperation' (language 'en')
== Spawn extension (n01, s, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
I also tried Background(thank-you-cooperation|m||n01). The result is
- pressing 1 goto ext i
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation)
in new stack
-- Playing 'thank-you-cooperation' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation|m||n01)
in new stack
-- Playing 'thank-you-cooperation' (language '')
-- Sent into invalid extension 'E8' in context 'n01' on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Playback(Zap/1-1, digits/star) in new
stack
-- Playing 'digits/star' (language 'en')
== Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'
NOTICE that * tries to go to ext 'E8' which is a French alphabet e with
grave accent.
DTMF detection problem? but if context option and m option of Background is
not specified, everything works well.
any help will be appreciated.
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[Asterisk-Users] can automon work with MixMonitor
Hi list automon now works as Monitor does. But MixMonitor is a better way for most cases, I guess. Any workaround to make automon do that? Any help will be appreciated. Franz Wu ___ --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
[Asterisk-Users] Dial cmd has delay for the last dialed number on FXO
Hi list
The * server one TDM04B card and my dialplan:
exten = 080.,1,Dial(Zap/g1/${EXTEN})
All four FXO ports has group=1 in zapata.conf
After dialing 0800012345 from a FXS extension, with one DTMF detector tapped
on the line, I found * dialed 0,8,0,0,0,1,2,3,4 in even interval, delay
longer, and then the last 5. Sometimes this behavior causes PSTN to repsond
with the number you dialed is non-existing.
Any help will be appreciated.
Franz
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[Asterisk-Users] option r in Dial command seems not to work
hi list need your opinion. thanks in advance when calls come in from zap channel (te110p as E1/PRI) and go out to a SIP peer, no ringtone heard at zap channel. sip.conf: [from_sipproxy] progressinband=no type=peer context=from-proxy host=xxx.yyy.zzz.www port=5060 disallow=all allow=alaw extensions.conf [pri] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,r) whether using option r in Dial command or not makes no diference. calls from sip to zap channel work well. i traced sip packets, found that sip device did send 180 Ringing. it looks like asterisk tells zap channel to say there's inband information but not to generate ringtone. on * console, pri debug span 1 shows Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 320/0x140) (Originator) Message type: SETUP (5) [a1]01*CLI Sending Complete (len= 1) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] [6c 0c 00 80 30 39 35 35 36 38 39 34 35 36] Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number not screened (0) '0955689456' ] [70 0b 80 30 37 30 32 30 33 34 31 30 30] Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '0702034100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4) -- Making new call for cr 320 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) -- Processing IE 125 (cs0, High-layer Compatibility) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 320/0x140) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] -- Executing Dial(Zap/10-1, SIP/[EMAIL PROTECTED]) in new stack -- Accepting call from '0955689456' to '0702034100' on channel 0/10, span 1 -- Called [EMAIL PROTECTED] -- SIP/220.228.43.182-9495 is ringing Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 320/0x140) (Terminator) Message type: ALERTING (1) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 320/0x140) (Originator) Message type: DISCONNECT (69) [08 02 80 90] ___ --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
Re: [Asterisk-Users] How to check Digium TE410P firmware version?
I connect to Asterisk via SSH all the times. Did not notice about console messages about module loading. Thanks - Original Message - From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 10:24 Subject: Re: [Asterisk-Users] How to check Digium TE410P firmware version? On 12/26/05, Franz Wu [EMAIL PROTECTED] wrote: Hi list I have one TE410P and want to know how to. Sending back to Digium should be a good idea. When you load the wct4xxp module you'll see (1st Gen) or (2nd Gen) pop up which will indicate which version of the firmware the board is. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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 ___ --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
[Asterisk-Users] How to check Digium TE410P firmware version?
Hi list I have one TE410P and want to know how to. Sending back to Digium should be a good idea. thanks in advance ___ --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
[Asterisk-Users] TDMoE problem
Hi all my system 1: celeron 1.2GHz + intel 810e (asus TUW-LA) + 256MB SDRAM onboard vga (intel 810e chipset) RTL8100 NIC debian sarge 3.1r0a / kernel 2.6.8-2-686 asterisk / libpri / zaptel from CVS HEAD @ 2005-10-24 system 2: pentium II 533MHz + intel 810e (dfi PW35-E) + 256MB SDRAM onboard vga RTL8139D NIC TE110P debian sarge 3.1r0a / kernel 2.6.8-2-686 asterisk / libpri / zaptel from CVS HEAD @ 2005-10-24 CASE 1: configuring TDMoE and TE100P as the same time make kernel hang zaptel.conf of system 1 dynamic=eth,eth0/MAC#2,31,1 bchan=1-15,17-31 dchan=16 zapata.conf context = p1 swtichtype = qsig signalling = pri_cpe resetinterval = 3600 channel = 1-15 channel = 17-31 zaptel.conf of system 2 # for TDMoE dynamic=eth,eth0/MAC#1,31,0 bchan=1-15,17-31 dchan=16 # for TE110P span=1,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 zapata.conf context = e1 swtichtype = qsig signalling = pri_cpe resetinterval = 3600 channel = 1-15 channel = 17-31 context=p2 swtichtype = qsig signalling = pri_net resetinterval = 3600 channel = 32-46 channel = 48-62 with this configuration, modprobe ok. ztcfg - ok. but at system boot-up, the kernel dumps a lot of garbage (softirq.c badness). if the TE110P card removed from PCI slot of system 2 as well as corresponding config, things go well as long as * does not start. CASE II TDMoE span has a lot of frame reject and D-channel down and up all configuration same as CASE I except the TE110P removed. when more than 5 calls are set up between two systems, messages about frame reject, PRI got event: HDLC Bad FCS dumps on console and D-channel looks like going up and down and up and down. even with 5 or less calls, the sound quality is bad. On the voip-info.org wiki, somebody seems having same problem as me. Any opinion will be appreciated. Franz Wu ___ --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] TDMoE and Badness in Kernel
2.6.13.4 which digium staff recommended. 2.6.14 both fail ___ --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
