RE: [Asterisk-Users] TDM400P FXO channel hookstate always "Offhook" &outbound digits sent before provider dialtone
Manny,
You are misinformed. My instructions were for [EMAIL PROTECTED]/AMP. If you
opent he
configuration page for the ZAP trunk, you simply put "ww" as the
Outbound Dial Prefix, save the settings, and clikc in the red banner to
reload.
Manual changes only get overwritten if you make them by hand to the
extensions.conf or extensions_additional.conf. If you add your changes
to the extensions_custom.conf, they will not be overwritten by AMP.
(That's how I force my incoming Zap trunk to ring to a different place
than my incoming call settings.)
Tom
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Manny A. Wise
> Sent: Friday, August 12, 2005 5:16 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] TDM400P FXO channel hookstate
> always "Offhook" &outbound digits sent before provider dialtone
>
>
> That's a good advice, BUT
> Remember we talking [EMAIL PROTECTED] here, it will get overwritten
> every time you do a configuration reload... That is what I
> did for my cellsocket, but guess what, I had to fix it every
> time I mess around with the system even adding an extension,
> it was s annoying, that now I have two systems running,
> one with [EMAIL PROTECTED] and another
> plain *... ;) Heck at least I learn to deal with both now..
> jejejejeje
>
> Manny
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Tom Rymes
> Sent: Friday, August 12, 2005 4:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] TDM400P FXO channel hookstate
> always "Offhook" &outbound digits sent before provider dialtone
>
> Open the [EMAIL PROTECTED] AMP interface, click on trunks, and click on the
> entry
> for your ZAP trunk. Then, put 'ww' in the "Outbound Dial Prefix"
>
> Tom
>
> On Aug 12, 2005, at 1:29 PM, Stephen Joyce wrote:
>
> > I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently
> > TDM400P with (1) FXO card on port 4. Inbound calls are always
> > but outbound calls fail 75% of the time with intercept messages
> > dial tone provider that include "we're sorry, your call did not go
> > through", and "we're sorry, when placing a local call it is now
> > necessary to dial an area code and the 7-digit number".
> > I have connected a test set in monitor mode to the phone line to
> > to what's being sent out the line by the Zap channel and 10
> digits are
> > sent but the first digit is usually sent only as I hear the
> dial tone
> > being drawn from the line, so it appears that it's sent before the
> > provider is ready to receive it. I can't find any sort of
> setting that
> > would allow me to manually configure a dialing delay on the line,
> > suspect this would provide a band-aid.
> > When looking at the Asterisk CLI, I see that the correct number is
> > dialed by my dial plan. I am calling from SIP extension 1100 and
> > 770-555-1234, which is a local 10-digit phone number.
> > -- Executing Dial("SIP/1100-9adc", "ZAP/g0/7705551234") in new stack
> > -- Called g0/7705551234
> > -- Zap/4-1 answered SIP/1100-9adc
> > The status of the channel is "Offhook" regardless of
> whether or not
> > phone line is actually Offhook or completely idle. I'm assuming that
> > when the line seems idle, it should show as Onhook.
>
>
> ___
> 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] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
Rich Adamson wrote: I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area code and the 7-digit number". Add a "w" in the dial string within extensions.conf. The "w" adds a short delay before sending the dtmf to the telco, which is likely happening too fast for the telco switch. Is that undocumented? The wiki entry for the Dial() command says: w: Allow the /called/ user to start recording after pressing *1 or what defined in features.conf (Asterisk > v1.0.x) To add a short delay to one of my remote FXO lines, I added A(silence/2) to the dial string. I don't recall seeing any changes come through for that, and "w" does add the delay before sending dtmf. Lots of folks use it, so I'd have to guess that either the 'show application dial' displays the help for w used in a different context. Embedded the "w" in the called number, and it will add about 200 milliseconds of delay. Multiple w's add more delay. I believe that the Dialplan option 'w' and the dial *string* meta-digit 'w' are the source of confusion here. They are distinct. One of them is found in the string of "digits" to be dialed and causes a wait; the other one is found after the "|" that separates the dial target from the options to the command. B. ___ 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] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
> >>I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently added a > >>TDM400P with (1) FXO card on port 4. Inbound calls are always successful > >>but outbound calls fail 75% of the time with intercept messages from my > >>dial tone provider that include "we're sorry, your call did not go > >>through", and "we're sorry, when placing a local call it is now > >>necessary to dial an area code and the 7-digit number". > >> > >> > > > >Add a "w" in the dial string within extensions.conf. The "w" adds a short > >delay before sending the dtmf to the telco, which is likely happening > >too fast for the telco switch. > > > > > Is that undocumented? > The wiki entry for the Dial() command says: > w: Allow the /called/ user to start recording after pressing *1 or what > defined in features.conf (Asterisk > v1.0.x) > > To add a short delay to one of my remote FXO lines, I added A(silence/2) > to the dial string. I don't recall seeing any changes come through for that, and "w" does add the delay before sending dtmf. Lots of folks use it, so I'd have to guess that either the 'show application dial' displays the help for w used in a different context. Embedded the "w" in the called number, and it will add about 200 milliseconds of delay. Multiple w's add more delay. If you use google to search the archives, you find lots of references to this. ___ 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] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
Rich Adamson wrote: I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area code and the 7-digit number". Add a "w" in the dial string within extensions.conf. The "w" adds a short delay before sending the dtmf to the telco, which is likely happening too fast for the telco switch. Is that undocumented? The wiki entry for the Dial() command says: w: Allow the /called/ user to start recording after pressing *1 or what defined in features.conf (Asterisk > v1.0.x) To add a short delay to one of my remote FXO lines, I added A(silence/2) to the dial string. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ 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] TDM400P FXO channel hookstate always "Offhook" &outbound digits sent before provider dialtone
That's a good advice, BUT
Remember we talking [EMAIL PROTECTED] here, it will get overwritten every time
you do a
configuration reload...
That is what I did for my cellsocket, but guess what, I had to fix it every
time I mess around with the system even adding an extension, it was s
annoying, that now I have two systems running, one with [EMAIL PROTECTED] and
another
plain *... ;) Heck at least I learn to deal with both now.. jejejejeje
Manny
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: Friday, August 12, 2005 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM400P FXO channel hookstate always "Offhook"
&outbound digits sent before provider dialtone
Open the [EMAIL PROTECTED] AMP interface, click on trunks, and click on the
entry
for your ZAP trunk. Then, put 'ww' in the "Outbound Dial Prefix"
Tom
On Aug 12, 2005, at 1:29 PM, Stephen Joyce wrote:
> I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently
> TDM400P with (1) FXO card on port 4. Inbound calls are always
> but outbound calls fail 75% of the time with intercept messages
> dial tone provider that include "we're sorry, your call did not go
> through", and "we're sorry, when placing a local call it is now
> necessary to dial an area code and the 7-digit number".
> I have connected a test set in monitor mode to the phone line to
> to what's being sent out the line by the Zap channel and 10 digits are
> sent but the first digit is usually sent only as I hear the dial tone
> being drawn from the line, so it appears that it's sent before the
> provider is ready to receive it. I can't find any sort of setting that
> would allow me to manually configure a dialing delay on the line,
> suspect this would provide a band-aid.
> When looking at the Asterisk CLI, I see that the correct number is
> dialed by my dial plan. I am calling from SIP extension 1100 and
> 770-555-1234, which is a local 10-digit phone number.
> -- Executing Dial("SIP/1100-9adc", "ZAP/g0/7705551234") in new stack
> -- Called g0/7705551234
> -- Zap/4-1 answered SIP/1100-9adc
> The status of the channel is "Offhook" regardless of whether or not
> phone line is actually Offhook or completely idle. I'm assuming that
> when the line seems idle, it should show as Onhook.
___
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] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
Open the [EMAIL PROTECTED] AMP interface, click on trunks, and click on the entry
for your ZAP trunk. Then, put 'ww' in the "Outbound Dial Prefix"
Tom
On Aug 12, 2005, at 1:29 PM, Stephen Joyce wrote:
I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently
added a
TDM400P with (1) FXO card on port 4. Inbound calls are always
successful
but outbound calls fail 75% of the time with intercept messages
from my
dial tone provider that include "we're sorry, your call did not go
through", and "we're sorry, when placing a local call it is now
necessary to dial an area code and the 7-digit number".
I have connected a test set in monitor mode to the phone line to
listen
to what's being sent out the line by the Zap channel and 10 digits are
sent but the first digit is usually sent only as I hear the dial tone
being drawn from the line, so it appears that it's sent before the
provider is ready to receive it. I can't find any sort of setting that
would allow me to manually configure a dialing delay on the line,
but I
suspect this would provide a band-aid.
When looking at the Asterisk CLI, I see that the correct number is
being
dialed by my dial plan. I am calling from SIP extension 1100 and
calling
770-555-1234, which is a local 10-digit phone number.
-- Executing Dial("SIP/1100-9adc", "ZAP/g0/7705551234") in new stack
-- Called g0/7705551234
-- Zap/4-1 answered SIP/1100-9adc
The status of the channel is "Offhook" regardless of whether or not
the
phone line is actually Offhook or completely idle. I'm assuming that
when the line seems idle, it should show as Onhook.
asterisk2*CLI> zap show channel 4
Channel: 4CLI>
File Descriptor: 15
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID string:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Offhook
Contents of /etc/zaptel.conf
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
# channel 1, WCTDM, inactive.
# channel 2, WCTDM, inactive.
# channel 3, WCTDM, inactive.
fxsks=4
# Global data
loadzone= us
defaultzone= us
Contents of /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=12.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include AMP configs
#include zapata_additional.conf
;Include genzaptelconf configs
#include zapata-auto.conf
Contents of /etc/asterisk/Zapata-auto.conf
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended
; to be #include-d by /etc/zapata.conf that will include the global
settings
;
; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
; channel 1, WCTDM, inactive.
; channel 2, WCTDM, inactive.
; channel 3, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from-pstn
group=0
channel => 4
-
Thanks for your consideration in reviewing my configuration and
suggesting some diagnostic steps and/or solutions.
Cheers - Stephen Joyce
___
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] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
Jeremy Gault wrote:
I'm not familiar with [EMAIL PROTECTED] itself but I do know what you probably need
in order to fix this. Somehow you will need to modify your dialplan
to include a "w" or two before the number. For example, instead of
dialing ZAP/g0/7705551234, dial ZAP/g0/ww7705551234 instead. Each "w"
inserts a 0.5 second pause, so two of them would give 1 second of
pause before dialing. This will give the outbound line time for a
dialtone to appear before your system starts dialing digits.
If someone else here knows how to tweak the [EMAIL PROTECTED] dialplans to implement
this, maybe they can reply and let you know how.
Jeremy
Stephen Joyce wrote:
I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently added a
TDM400P with (1) FXO card on port 4. Inbound calls are always successful
but outbound calls fail 75% of the time with intercept messages from my
dial tone provider that include "we're sorry, your call did not go
through", and "we're sorry, when placing a local call it is now
necessary to dial an area code and the 7-digit number".
I have connected a test set in monitor mode to the phone line to listen
to what's being sent out the line by the Zap channel and 10 digits are
sent but the first digit is usually sent only as I hear the dial tone
being drawn from the line, so it appears that it's sent before the
provider is ready to receive it. I can't find any sort of setting that
would allow me to manually configure a dialing delay on the line, but I
suspect this would provide a band-aid.
When looking at the Asterisk CLI, I see that the correct number is being
dialed by my dial plan. I am calling from SIP extension 1100 and calling
770-555-1234, which is a local 10-digit phone number.
-- Executing Dial("SIP/1100-9adc", "ZAP/g0/7705551234") in new stack
-- Called g0/7705551234
-- Zap/4-1 answered SIP/1100-9adc
The status of the channel is "Offhook" regardless of whether or not the
phone line is actually Offhook or completely idle. I'm assuming that
when the line seems idle, it should show as Onhook.
asterisk2*CLI> zap show channel 4
Channel: 4CLI>
File Descriptor: 15
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID string:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Offhook
Contents of /etc/zaptel.conf
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" # channel 1,
WCTDM, inactive.
# channel 2, WCTDM, inactive.
# channel 3, WCTDM, inactive.
fxsks=4
# Global data
loadzone= us
defaultzone= us
Contents of /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=12.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include AMP configs
#include zapata_additional.conf
;Include genzaptelconf configs
#include zapata-auto.conf
Contents of /etc/asterisk/Zapata-auto.conf
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended ; to be #include-d by /etc/zapata.conf that will include the
global
settings
;
; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" ; channel 1,
WCTDM, inactive.
; channel 2, WCTDM, inactive.
; channel 3, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from-pstn
group=0
channel => 4
-
Thanks for your consideration in reviewing my configuration and
suggesting some diagnostic steps and/or solutions.
Cheers - Stephen Joyce
___
That would be Outbound Dial Prefix on the Trunks page in AAH/AMP
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailm
Re: [Asterisk-Users] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
> I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently added a > TDM400P with (1) FXO card on port 4. Inbound calls are always successful > but outbound calls fail 75% of the time with intercept messages from my > dial tone provider that include "we're sorry, your call did not go > through", and "we're sorry, when placing a local call it is now > necessary to dial an area code and the 7-digit number". Add a "w" in the dial string within extensions.conf. The "w" adds a short delay before sending the dtmf to the telco, which is likely happening too fast for the telco switch. ___ 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] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
I'm not familiar with [EMAIL PROTECTED] itself but I do know what you probably need in
order to fix this. Somehow you will need to modify your dialplan to
include a "w" or two before the number. For example, instead of dialing
ZAP/g0/7705551234, dial ZAP/g0/ww7705551234 instead. Each "w" inserts a
0.5 second pause, so two of them would give 1 second of pause before
dialing. This will give the outbound line time for a dialtone to appear
before your system starts dialing digits.
If someone else here knows how to tweak the [EMAIL PROTECTED] dialplans to implement
this, maybe they can reply and let you know how.
Jeremy
Stephen Joyce wrote:
I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently added a
TDM400P with (1) FXO card on port 4. Inbound calls are always successful
but outbound calls fail 75% of the time with intercept messages from my
dial tone provider that include "we're sorry, your call did not go
through", and "we're sorry, when placing a local call it is now
necessary to dial an area code and the 7-digit number".
I have connected a test set in monitor mode to the phone line to listen
to what's being sent out the line by the Zap channel and 10 digits are
sent but the first digit is usually sent only as I hear the dial tone
being drawn from the line, so it appears that it's sent before the
provider is ready to receive it. I can't find any sort of setting that
would allow me to manually configure a dialing delay on the line, but I
suspect this would provide a band-aid.
When looking at the Asterisk CLI, I see that the correct number is being
dialed by my dial plan. I am calling from SIP extension 1100 and calling
770-555-1234, which is a local 10-digit phone number.
-- Executing Dial("SIP/1100-9adc", "ZAP/g0/7705551234") in new stack
-- Called g0/7705551234
-- Zap/4-1 answered SIP/1100-9adc
The status of the channel is "Offhook" regardless of whether or not the
phone line is actually Offhook or completely idle. I'm assuming that
when the line seems idle, it should show as Onhook.
asterisk2*CLI> zap show channel 4
Channel: 4CLI>
File Descriptor: 15
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID string:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Offhook
Contents of /etc/zaptel.conf
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
# channel 1, WCTDM, inactive.
# channel 2, WCTDM, inactive.
# channel 3, WCTDM, inactive.
fxsks=4
# Global data
loadzone= us
defaultzone = us
Contents of /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=12.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include AMP configs
#include zapata_additional.conf
;Include genzaptelconf configs
#include zapata-auto.conf
Contents of /etc/asterisk/Zapata-auto.conf
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended
; to be #include-d by /etc/zapata.conf that will include the global
settings
;
; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
; channel 1, WCTDM, inactive.
; channel 2, WCTDM, inactive.
; channel 3, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from-pstn
group=0
channel => 4
-
Thanks for your consideration in reviewing my configuration and
suggesting some diagnostic steps and/or solutions.
Cheers - Stephen Joyce
___
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] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently added a
TDM400P with (1) FXO card on port 4. Inbound calls are always successful
but outbound calls fail 75% of the time with intercept messages from my
dial tone provider that include "we're sorry, your call did not go
through", and "we're sorry, when placing a local call it is now
necessary to dial an area code and the 7-digit number".
I have connected a test set in monitor mode to the phone line to listen
to what's being sent out the line by the Zap channel and 10 digits are
sent but the first digit is usually sent only as I hear the dial tone
being drawn from the line, so it appears that it's sent before the
provider is ready to receive it. I can't find any sort of setting that
would allow me to manually configure a dialing delay on the line, but I
suspect this would provide a band-aid.
When looking at the Asterisk CLI, I see that the correct number is being
dialed by my dial plan. I am calling from SIP extension 1100 and calling
770-555-1234, which is a local 10-digit phone number.
-- Executing Dial("SIP/1100-9adc", "ZAP/g0/7705551234") in new stack
-- Called g0/7705551234
-- Zap/4-1 answered SIP/1100-9adc
The status of the channel is "Offhook" regardless of whether or not the
phone line is actually Offhook or completely idle. I'm assuming that
when the line seems idle, it should show as Onhook.
asterisk2*CLI> zap show channel 4
Channel: 4CLI>
File Descriptor: 15
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID string:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Offhook
Contents of /etc/zaptel.conf
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
# channel 1, WCTDM, inactive.
# channel 2, WCTDM, inactive.
# channel 3, WCTDM, inactive.
fxsks=4
# Global data
loadzone= us
defaultzone = us
Contents of /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=12.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include AMP configs
#include zapata_additional.conf
;Include genzaptelconf configs
#include zapata-auto.conf
Contents of /etc/asterisk/Zapata-auto.conf
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended
; to be #include-d by /etc/zapata.conf that will include the global
settings
;
; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
; channel 1, WCTDM, inactive.
; channel 2, WCTDM, inactive.
; channel 3, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from-pstn
group=0
channel => 4
-
Thanks for your consideration in reviewing my configuration and
suggesting some diagnostic steps and/or solutions.
Cheers - Stephen Joyce
___
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
