[Asterisk-Users]Hangup with error

2006-03-08 Thread asterisk183
I used quadBri Junghanns card and I config
zaptel.conf:
ZAPTEL.CONF
loadzone=it
defaultzone=it
span=1,1,3,ccs,ami
span=2,2,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami

bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12

ZAPATA.CONF
[channels]
language=it
musiconhold=default
switchtype = euroisdn

; p2mp TE mode (for connecting ISDN lines in
point-to-multipoint mode)
signalling = bri_cpe_ptmp
; p2p TE mode (for connecting ISDN lines in
point-to-point mode)
;signalling = bri_cpe
; p2mp NT mode (for connecting ISDN phones in
point-to-multipoint mode)
;signalling = bri_net_ptmp
; p2p NT mode (for connecting an ISDN pbx in
point-to-point mode)
;signalling = bri_net

pridialplan = local
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00

echocancel = yes

context=isdn_incoming
group = 1
channel = 1-2

group = 2
channel = 4-5

group = 3
channel = 7-8

group = 4
channel = 10-11

But when I hangup the channel, Asterisk show this
message:

Mar  6 17:31:20 WARNING[1437]: chan_zap.c:6570
handle_init_event: Detected alarm on channel 1: Red
Alarm
Mar  6 17:31:20 WARNING[1437]: chan_zap.c:1593
zt_disable_ec: Unable to disable echo cancellation on
channel 1
Mar  6 17:31:20 WARNING[1437]: chan_zap.c:6570
handle_init_event: Detected alarm on channel 2: Red
Alarm
Mar  6 17:31:20 WARNING[1437]: chan_zap.c:1593
zt_disable_ec: Unable to disable echo cancellation on
channel 2
Mar  6 17:31:20 NOTICE[1433]: chan_zap.c:8511
pri_dchannel: PRI got event: Alarm (4) on Primary
D-channel of span 1
Mar  6 17:31:20 NOTICE[1433]: chan_zap.c:8518
pri_dchannel: pri_shutdown
Mar  6 17:31:20 NOTICE[1437]: chan_zap.c:6565
handle_init_event: Alarm cleared on channel 1
Mar  6 17:31:20 NOTICE[1437]: chan_zap.c:6565
handle_init_event: Alarm cleared on channel 2
Mar  6 17:31:20 NOTICE[1433]: chan_zap.c:8511
pri_dchannel: PRI got event: No more alarm (5) on
Primary D-channel of span 1

Why? And What i can doing for solve this problem?

Thanks






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[Asterisk-Users]chan_zap.c:6570 handle_init_event error

2006-03-06 Thread asterisk183
I used quadBri Junghanns card and I config zaptel.conf: ZAPTEL.CONF loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami  bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12  ZAPATA.CONF [channels] language=it musiconhold=default switchtype = euroisdn  ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode) ;signalling = bri_net_ptmp ; p2p NT mode (for connecting an ISDN pbx in point-to-point mode) ;signalling = bri_net  pridialplan = local prilocaldialplan = local nationalprefix = 0 internationalprefix = 00  echocancel = yes  context=isdn_incoming group
 = 1 channel = 1-2  group = 2 channel = 4-5  group = 3 channel = 7-8  group = 4 channel = 10-11  But when I hangup the channel, Asterisk show this message:  Mar 6 17:31:20 WARNING[1437]: chan_zap.c:6570 handle_init_event: Detected alarm on channel 1: Red Alarm Mar 6 17:31:20 WARNING[1437]: chan_zap.c:1593 zt_disable_ec: Unable to disable echo cancellation on channel 1 Mar 6 17:31:20 WARNING[1437]: chan_zap.c:6570 handle_init_event: Detected alarm on channel 2: Red Alarm Mar 6 17:31:20 WARNING[1437]: chan_zap.c:1593 zt_disable_ec: Unable to disable echo cancellation on channel 2 Mar 6 17:31:20 NOTICE[1433]: chan_zap.c:8511 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 Mar 6 17:31:20 NOTICE[1433]: chan_zap.c:8518 pri_dchannel: pri_shutdown Mar 6 17:31:20 NOTICE[1437]: chan_zap.c:6565 handle_init_event: Alarm cleared on
  channel
 1 Mar 6 17:31:20 NOTICE[1437]: chan_zap.c:6565 handle_init_event: Alarm cleared on channel 2 Mar 6 17:31:20 NOTICE[1433]: chan_zap.c:8511 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1  Why? And What i can doing for solve this problem?  Thanks 
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[Asterisk-Users] echo problem

2006-02-14 Thread asterisk183
I have installed Asterisk and when I hangup the zap channel Asterisk show this message:  Feb 13 17:45:49 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 4: Red AlarmFeb 13 17:45:49 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 4Feb 13 17:45:49 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 5: Red AlarmFeb 13 17:45:49 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 5Feb 13 17:45:49 NOTICE[1745]: chan_zap.c:8451 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 2Feb 13 17:45:49 NOTICE[1745]: chan_zap.c:8458 pri_dchannel: pri_shutdownFeb 13 17:45:49 NOTICE[1748]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 4Feb 13 17:45:49 NOTICE[1748]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 5Feb 1
 3
 17:45:49 NOTICE[1745]: chan_zap.c:8451 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 2  What I can doing?Thanks  
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[Asterisk-Users] problem with Zaptel

2006-02-07 Thread asterisk183
When I doing hangup the zaptel channel, Asterisk show this message:  Feb 7 12:14:44 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 4: Red Alarm Feb 7 12:14:44 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 4 Feb 7 12:14:44 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 5: Red Alarm Feb 7 12:14:44 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 5 Feb 7 12:14:44 NOTICE[1745]: chan_zap.c:8451 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 2 Feb 7 12:14:44 NOTICE[1745]: chan_zap.c:8458 pri_dchannel: pri_shutdown Feb 7 12:14:44 NOTICE[1748]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 4 Feb 7 12:14:44 NOTICE[1748]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 5 Feb 7 12:14:44 NOTICE[1745]:
 chan_zap.c:8451 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 2   Why? And What I can doing?  Thanks 
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Re: [Asterisk-Users] CallerID popup

2006-02-03 Thread asterisk183
Dial event, in Asterisk 1.2:203-201Event: Dial  Privilege: call,all  Source: SIP/203-8467  Destination: SIP/201-45d9  CallerID: 203  CallerIDName: 203  SrcUniqueID: asterisk-1912-1138197095.3769  DestUniqueID: asterisk-1912-1138197095.3771Mimmus [EMAIL PROTECTED] ha scritto:   Hi,I'm trying to write a small Visual Basic app to throw a popup withCallerIDNum when a call center agent answers a queue call.Does anyone know what is the right manager event to intercept?ThanksMimmus___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users]MusicOnHold don't start at begin

2006-01-04 Thread asterisk183
MusiconHold don't start at begin.  What I can doing for setup the musiconhold start at begin?Thanks  Fabio  
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[Asterisk-Users]CLI execute extensions

2005-12-28 Thread asterisk183
What I can doing execute extension (es.200) from CLI?  Thanks 
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[Asterisk-Users] Name file automatic

2005-12-21 Thread asterisk183
Hi to everybody, I have set up Asterisk that when I am in conversation pressing * 1 activates in an automatic way the recording of the call. (as expressed in feature.conf) the file which is registered is a wav file with a name produced in an automatic way by Asterisk. Is there the possibility to set up a different name and change the format of the recording?  Thanks Fabio 
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[Asterisk-Users] MixMonitor error exit

2005-12-19 Thread asterisk183
When I used MixMonitor, and I hangup the channel, Asterisk exit whit error in segment.  Why?  Thanks 
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[Asterisk-Users] CallerID Transfer

2005-12-12 Thread asterisk183
When I receveid a call (num1) in the my office (num2), I transfer the call at the num3, but the callerid is num2, in the telephone3. What can I doing for show the callerid num1?  Thanks 
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[Asterisk-Users] Call transfer error

2005-12-01 Thread asterisk183
When I arrived a call, I would the call transfer in to another telephone number, but Asterisk show error:  Executing GotoIfTime("Zap/4-1", "08:30-12:30|mon-fri|*|*?4") in new stack -- Executing GotoIfTime("Zap/4-1", "15:30-18:30|mon-fri|*|*?4") in new stack -- Executing Goto("Zap/4-1", "6") in new stack  -- Goto (isdn_incoming,0445363378,6)  -- Executing Dial("Zap/4-1", "ZAP/g2/0445384225|60") in new stack  -- Requested transfer capability: 0x00 - SPEECH  -- Called g2/0445384225  -- Zap/5-1 is proceeding passing it to Zap/4-1  -- Channel 0/2, span 2 got hangup request  -- Hungup 'Zap/5-1'  == No one is available to answer at this time (1:0/0/0)  -- Executing Hangup("Zap/4-1", "") in new stack  == Spawn exten
 sion
 (isdn_incoming, 0445363378, 7) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1'  MY ZAPATA.CONF IS: [channels] switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel = yes context=isdn_incoming group = 1 channel = 1-2 group = 2 channel = 4-5 group = 3 channel = 7-8 group = 4 channel = 10-11  MY ZAPTEL.CONF is loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12  MY EXTENSIONS.CONF is [isdn_incoming] exten = 0445363378,1,GotoIfTime(${ORAMATTINO}?4) exten = 0445363378,2,GotoIfTime(${ORAPOMERIGGIO}?4) exten = 0445363378,3,Goto(6) exten =
 0445363378,4,Dial(${TELEIN},60) exten = 0445363378,5,Hangup exten = 0445363378,6,Dial(ZAP/g2/0445384225,60) exten = 0445363378,7,Hangup  What can I doing?  Thanks 
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[Asterisk-Users] Help transfer call

2005-11-30 Thread asterisk183
I want to transfer a telephone call in a house number in determinate established hours, this syntax is correct or it must use a command different from DIAL?  exten = _x.,1,GotoIfTime(08:30-12:30|mon-fri|*|*?4) exten = _x.,2,GotoIfTime(14:30-18:30|mon-fri|*|*?4) exten = _x.,3,Goto(cellulare,s,1) exten = _x.,4,Dial(${TELEIN},60) exten = _x.,5,Hangup  [cellulare] exten = s,1,Dial(ZAP/g2/${TELETRASFERIMENTO},60) exten = s,2,Hangup  Thanks Ciao, Fabio 
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[Asterisk-Users] Tone busy in zaptel

2005-11-30 Thread asterisk183
I use the Zaptel card and when I call a client busy, Asterisk don't play standard tone of busy, but Asterisk play forbidden tone.  What can I doing for play busy tone to Asterisk?  Thanks Fabio 
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[Asterisk-Users] Transfer call error

2005-11-30 Thread asterisk183
When I call a internal Sip telephone, the calling transfer to Teleohone external, but Asterisk show this error:  Executing Dial("SIP/201-1e2a", "ZAP/g1/3472543320|60") in new stack  -- Requested transfer capability: 0x00 - SPEECH  -- Called g1/3472543320 Nov 30 15:52:09 WARNING[1866]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 2: Red Alarm Nov 30 15:52:09 WARNING[1866]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 2 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8451 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8458 pri_dchannel: pri_shutdown Nov 30 15:52:09 NOTICE[1866]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 2 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8451 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 Nov 30 
 15:52:09
 WARNING[1866]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 1: No Alarm Nov 30 15:52:09 WARNING[1866]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 1 Nov 30 15:52:09 NOTICE[1866]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 1  -- Hungup 'Zap/1-1'  == No one is available to answer at this time (1:0/0/0)  -- Executing Hangup("SIP/201-1e2a", "") in new stack  == Spawn extension (local, 203, 2) exited non-zero on 'SIP/201-1e2a'   My extension.conf:  exten = 203,1,Dial(ZAP/g1/3472543320,60) exten = 203,2,Hangup  Why?  Thanks  
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[Asterisk-Users] chan_sip.c error

2005-11-30 Thread asterisk183
Why Asterisk show this message: Nov 30 17:05:17 WARNING[1351]: chan_sip.c:9600 handle_response_register: Got 200 OK on REGISTER that isn't a register  Thanks 
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[Asterisk-Users] qozap.o error

2005-11-29 Thread asterisk183
I am trying to install the qozap driver, but when I doing:  make all  the shell command show error in qozap.o.What can I doing for compiling qozap.o?Thanks  
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Re: [Asterisk-Users] qozap.o error

2005-11-29 Thread asterisk183
I risolved my problem: I have kernel source in  /usr/src/linux-2.4.686 instead of /usr/src/linux, therefore the qozap.c  doesn't compiling.ThanksTzafrir Cohen [EMAIL PROTECTED] ha scritto:   On Tue, Nov 29, 2005 at 03:56:24PM +0100, asterisk183 wrote:  I am trying to install the qozap driver, but when I doing:   make all   the shell command show error in qozap.o.  What can I doing for compiling qozap.o?  ThanksStart by giving the telepathy-chalanged among us some clues of yoursetup:- version of asterisk- version of bristuff- Linux version- what did you do so far?I should note that for Debian Sarge I already have them pre-built andpackaged. But then again, you probably don't want to miss the
 fun.-- Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il |   | a Mutt's  [EMAIL PROTECTED] |   |  bestICQ# 16849755 |   | friend___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Bristuff: qozap.o error

2005-11-25 Thread asterisk183
I have installed bristuff 0.3.0 for Asterisk 1.2 with kernel 2.4, but when I doing : insmod qozap.o  the shell show this messagge: qozap.o: qozap.o: unresolved symbol free_irq_Rsmp_f20dabd8 qozap.o: qozap.o: unresolved symbol pci_find_device_Rsmp_c584f4e3 qozap.o: qozap.o: unresolved symbol __request_region_Rsmp_1a1a4f09 qozap.o: qozap.o: unresolved symbol pci_disable_device_Rsmp_95846005 qozap.o: qozap.o: unresolved symbol iounmap_Rsmp_5fb196d4 qozap.o: qozap.o: unresolved symbol pci_enable_device_Rsmp_1bc741d2 qozap.o: qozap.o: unresolved symbol sprintf_Rsmp_1d26aa98 qozap.o: qozap.o: unresolved symbol __ioremap_Rsmp_9eac042a qozap.o: qozap.o: unresolved symbol pci_write_config_word_Rsmp_f23d8795 qozap.o: qozap.o: unresolved symbol printk_Rsmp_1b7d4074 qozap.o: qozap.o: unresolved symbol kfree_Rsmp_037a0cba qozap.o: qozap.o: unresolved symbol ioport_resource_Rsmp_865ebccd qozap.o: qoz
 ap.o:
 unresolved symbol kmalloc_Rsmp_93d4cfe6 qozap.o: qozap.o: unresolved symbol request_irq_Rsmp_0c60f2e0 qozap.o: qozap.o: unresolved symbol __release_region_Rsmp_d49501d4  What I can doing for resolving this problem? I trying copy qozap.o in /lib/modules/`uname -r`/misc/ but don't resolve the problem  Help  Thanks 
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[Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error

2005-11-24 Thread asterisk183
I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the instruction in INSTALL:  1. make menuconfig  2. make dep  3. ./install.sh  4. copy the file zapata.cong and zaptel.conf  5. modprobe zaptel  6.   But when I doing insmod qozap.o   and ztcfg don't start because in /qozap directory I don't have qozap.o files. Why?I have a quadBRI cardHELP!!!Thanks
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Re: [Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error

2005-11-24 Thread asterisk183
Show:  insmod: qozap.o: No such file or directoryKristof Hardy [EMAIL PROTECTED] ha scritto:   asterisk183 wrote: 5. modprobe zaptel 6.   But when I doing insmod qozap.o   and ztcfg don't start because in /qozap directory I don't have qozap.o  files. Why?what is the output you get 'after' you do: modprobe qozapAfter this, what is the output you get after: ztcfg -vAlso, the qozap.ko file should have been copied to your /lib/modules//misc directory. (this is being done by the install script)Cheers___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing
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Re: [Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error

2005-11-24 Thread asterisk183
kernel 2.4  Tzafrir Cohen [EMAIL PROTECTED] ha scritto:  On Thu, Nov 24, 2005 at 11:28:00AM +0100, asterisk183 wrote:  I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the instruction in INSTALL:   1. make menuconfig   2. make dep   3. ./install.sh   4. copy the file zapata.cong and zaptel.conf   5. modprobe zaptel   6. But when I doing insmod qozap.o and ztcfg don't start because in /qozap directory I don't have qozap.o files. Why?  I have a quadBRI cardkernel 2.4 or 2.6? On 2.6 you should have qozap.ko-- Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il |   | a Mutt's  [EMAIL PROTECTED] | 
 
  |  bestICQ# 16849755 |   | friend___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Don't Outgoing call with Zap

2005-11-24 Thread asterisk183
I have a QuadBRI card installed, and I received the call incoming, but I don't place call outgoing.  Asterisk show this message:  Executing Dial("SIP/101-a440", "ZAP/g1/3472543320|60") in new stack  -- Requested transfer capability: 0x00 - SPEECH  -- Called g1/3472543320 Nov 24 15:43:14 WARNING[10258]: chan_zap.c:6511 handle_init_event: Detected alar m on channel 2: Red Alarm Nov 24 15:43:14 WARNING[10258]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 2 Nov 24 15:43:14 NOTICE[10254]: chan_zap.c:8451 pri_dchannel: PRI got event: Alar m (4) on Primary D-channel of span 1 Nov 24 15:43:14 NOTICE[10254]: chan_zap.c:8458 pri_dchannel: pri_shutdown Nov 24 15:43:14 NOTICE[10258]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 2 Nov 24 15:43:14 NOTICE[10254]: chan_zap.c:8451 pri_dchannel: PRI got event: No m ore alarm (5) on
  Primary
 D-channel of span 1 Nov 24 15:43:14 WARNING[10258]: chan_zap.c:6511 handle_init_event: Detected alar m on channel 1: No Alarm Nov 24 15:43:14 WARNING[10258]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 1 Nov 24 15:43:14 NOTICE[10258]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 1  -- Hungup 'Zap/1-1'  == No one is available to answer at this time (1:0/0/0)  -- Executing Hangup("SIP/101-a440", "") in new stack  == Spawn extension (local, 3472543320, 2) exited non-zero on 'SIP/101-a440'  Why?  My extension.conf is: [general] static=yes writeprotect=no  [globals] TELEIN=SIP/200  ; ; CONTESTO PER LE CHIMATE IN INGRESSO LOCALI E IN USCITA * ; [local] <
 br>
 exten = _x.,1,Dial(ZAP/g1/${EXTEN},60) exten = _x.,2,Hangup   ;** ; CONTESTO PER LE CHIAMATE IN INGRESSO DA UNIVOICE * ;** [out-sip] exten = 101,1,Dial(${TELEIN},20,rt) exten = 101,3,Hangup  [isdn_incoming] exten = _x.,1,Dial(SIP/200,60) exten = _x.,2,Hangup   My zapata.conf is  [channels] switchtype = euroisdn  signalling = bri_cpe_ptmp pridialplan = local prilocaldialplan = local nationalprefix = 0 internationalprefix = 00  echocancel = yes  context=isdn_incoming group = 1 channel = 1-2 group = 2 channel = 4-5  Thanks 
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[Asterisk-Users] Asterisk pop-up

2005-11-15 Thread asterisk183
I search a application pop-up for the call.  When I recevea call, I want a pop-up menu. Where I can found this application open source?Thanks  Fabio
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[Asterisk-Users] [Asterisk-User] Festival help

2005-11-09 Thread asterisk183
I have install Festival in Asterisk, but I don't
listen to anything, but Asterisk show this message

Parsing '/etc/asterisk/cdr_custom.conf': Found
-- Executing Answer(SIP/101-35e3, ) in new
stack
-- Executing Festival(SIP/101-35e3, Hello
asterisk user| how are you today?) in new stack
  == Parsing '/etc/asterisk/festival.conf': Found

therefore don't show error.

Why?


Thanks






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Re: RE : RE : [Asterisk-Users] codecs

2005-11-09 Thread asterisk183
If you want convert file audio, you using this on line
apllication:
http://www.asteriskguru.com/tools/audio_conversion.php



--- Eric \ManxPower\ Wieling [EMAIL PROTECTED] ha
scritto: 

 Olivier Taylor wrote:
  User-agents have g729, g723.1 and gsm, isn't it
 possible to force user-agent
  to use gsm for voicemail and g729 for outbound
 calls?
 
 No.
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