[asterisk-users] Asterisk Died - Ver-1.6.2.6.

[Nitesh Divecha]

Hello All,

"safe_asterisk" just sent me an email saying "Asterisk on bill exited on 
signal 11.  Might want to take a peek.". Looking at the 
/var/log/asterisk/message doesn't show me anything...

This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and 
it is routing calls from Nextone MSW Softswitch to VPS Softswitch...

Any reason why Asterisk died?

Thanking in advance...

Cheers,
Nitesh


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[asterisk-users] Change SIP Release Code

[Nitesh Divecha]

Hello All,

I have configured Asterisk to act like a Softswitch (routing calls in 
and out) but I m facing one issue... I want to route advance all fail 
calls with ISDN 34 (SIP 503) whenever calls are failed due to some 
reasons...

Is there any way to flash back SIP 503 on all failed calls from Asterisk?

Cheers,
Nitesh

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Re: [asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile

[Nitesh Divecha]

Thanks Tzafrir

Kinda solve the issue by commenting out following: -

CHKCONFIG   := $(wildcard /sbin/chkconfig)
UPDATE_RCD  := $(wildcard /usr/sbin/update-rc.d)
ifeq (,$(DESTDIR))
#  ifneq (,$(CHKCONFIG))
#ADD_INITD  := $(CHKCONFIG) --add dahdi
#  else
ifneq (,$(UPDATE_RCD))
  ADD_INITD := $(UPDATE_RCD) dahdi defaults 15 30
endif
#  endif
endif

I don't know if this will cause any Asterisk installation... I m not 
going to use any Dahdi hardware...

Cheers,
Nitesh


Tzafrir Cohen wrote:
> On Sun, Mar 14, 2010 at 01:32:09PM -0400, Nitesh Divecha wrote:
>   
>> Hello All,
>>
>> I'm trying to do a fresh installation on Ubuntu Server 9.10 (Karmic) 
>> 64-bit but I am getting error when "make config" is trying to install 
>> the init script... Here is the output: - Can anyone help me please... 
>> Thanking in advance...
>> 
>
> The dahdi-tools Makefile tries to use chkconfig first before update-rc.d .
>
> On your system chkconfig actually exists, but it uses insserv (with
> service dependencies) and aparantly you have many services on your
> system without proper depdendencies declared. Unless those are old and
> unpurged ifles from obsolete packages.
>
> In short: I'm not sure what's the proper fix here (as opposed to the
> workaround of avoiding chkconfig .
>
>   


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Re: [asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile

[Nitesh Divecha]

e2 at depth 3
insserv:  loop involving service rsyslog at depth 2
insserv:  loop involving service udev at depth 1
insserv: There is a loop between service rsyslog and apache2 if stopped
insserv:  loop involving service mysql at depth 2
insserv:  loop involving service hwclock at depth 3
insserv: exiting without changing boot order!
/sbin/insserv failed, exit code 1
dahdi 0:off  1:off  2:off  3:off  4:off  5:off  6:off
make[1]: *** [config] Error 1
make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools'
make: *** [config] Error 2



tjoen wrote:
> On Sun, 2010-03-14 at 13:32 -0400, Nitesh Divecha wrote:
>
>   
>> make[1]: Entering directory 
>> 
> ...
>   
>> /sbin/chkconfig --add dahdi
>> insserv: warning: script 'S20theserver' missing LSB tags and overrides
>> 
> [snip other warnings]
>   
>> insserv: There is a loop between service rsyslog and apache2 if stopped
>> insserv:  loop involving service apache2 at depth 3
>> insserv:  loop involving service rsyslog at depth 2
>> insserv:  loop involving service udev at depth 1
>> insserv: There is a loop between service rsyslog and apache2 if stopped
>> insserv:  loop involving service mysql at depth 2
>> insserv:  loop involving service hwclock at depth 3
>> 
>
> The error?
>
>   
>> insserv: exiting without changing boot order!
>> /sbin/insserv failed, exit code 1
>> dahdi 0:off  1:off  2:off  3:off  4:off  5:off  6:off
>> make[1]: *** [config] Error 1
>> make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools'
>> make: *** [config] Error 2
>> 
>
> Edit /usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools/Makefile
> so it doesn't run /sbin/chkconfig
>
>
>
>   


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[asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile

[Nitesh Divecha]

Hello All,

I'm trying to do a fresh installation on Ubuntu Server 9.10 (Karmic) 
64-bit but I am getting error when "make config" is trying to install 
the init script... Here is the output: - Can anyone help me please... 
Thanking in advance...

Cheers,
Nitesh

###
###
### DAHDI tools installed successfully.
### If you have not done so before, install init scripts with:
###
###   make config
###
###
make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools'
make -C tools config
make[1]: Entering directory 
`/usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools'
install -D dahdi.init /etc/init.d/dahdi
/sbin/chkconfig --add dahdi
insserv: warning: script 'S20theserver' missing LSB tags and overrides
insserv: warning: current start runlevel(s) (0 6) of script `umountfs' 
overwrites defaults (empty).
insserv: warning: current start runlevel(s) (0 6) of script `umountroot' 
overwrites defaults (empty).
insserv: warning: script 'dbus' missing LSB tags and overrides
insserv: warning: script 'apport' missing LSB tags and overrides
insserv: warning: script 'udev-finish' missing LSB tags and overrides
insserv: warning: current start runlevel(s) (0 6) of script 
`umountnfs.sh' overwrites defaults (empty).
insserv: warning: script 'failsafe-x' missing LSB tags and overrides
insserv: warning: current start runlevel(s) (0 6) of script `sendsigs' 
overwrites defaults (empty).
insserv: warning: script 'udevtrigger' missing LSB tags and overrides
insserv: warning: current start runlevel(s) (0 6) of script 
`wpa-ifupdown' overwrites defaults (empty).
insserv: warning: script 'udevmonitor' missing LSB tags and overrides
insserv: warning: script 'udev' missing LSB tags and overrides
insserv: warning: script 'atd' missing LSB tags and overrides
insserv: warning: script 'dmesg' missing LSB tags and overrides
insserv: warning: script 'rsyslog-kmsg' missing LSB tags and overrides
insserv: warning: script 'procps' missing LSB tags and overrides
insserv: warning: script 'module-init-tools' missing LSB tags and overrides
insserv: warning: current start runlevel(s) (0 6) of script `networking' 
overwrites defaults (empty).
insserv: warning: script 'ufw' missing LSB tags and overrides
insserv: warning: script 'hwclock' missing LSB tags and overrides
insserv: warning: script 'theserver' missing LSB tags and overrides
insserv: warning: script 'cron' missing LSB tags and overrides
insserv: warning: current start runlevel(s) (0) of script `halt' 
overwrites defaults (empty).
insserv: warning: current start runlevel(s) (6) of script `reboot' 
overwrites defaults (empty).
insserv: warning: script 'rsyslog' missing LSB tags and overrides
insserv: warning: script 'hwclock-save' missing LSB tags and overrides
insserv: There is a loop between service rsyslog and apache2 if stopped
insserv:  loop involving service apache2 at depth 3
insserv:  loop involving service rsyslog at depth 2
insserv:  loop involving service udev at depth 1
insserv: There is a loop between service rsyslog and apache2 if stopped
insserv:  loop involving service mysql at depth 2
insserv:  loop involving service hwclock at depth 3
insserv: exiting without changing boot order!
/sbin/insserv failed, exit code 1
dahdi 0:off  1:off  2:off  3:off  4:off  5:off  6:off
make[1]: *** [config] Error 1
make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools'
make: *** [config] Error 2


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Re: [asterisk-users] TDM410P - False Answer Supervision

[Nitesh Divecha]

Danny,

Thanks for your reply...

ForkCDR will take care the problem locally but what about the originator 
of the call? Calls are originated from the Nextone Softswitch (Customer) 
to Asterisk and while the call is still in progress Asterisk has already 
sent back "Answered" and the call is accounted Normal code 16 on Nextone 
Billing (Customer). So all the time ASR is at 100% from the 
originator...  which is an issue we are facing...

Cheers,
Nitesh


Danny Nicholas wrote:
> Since POTS supervision is questionable at best, the best option IMO would be
> to use something like ForkCDR to let you know that the call has really been
> answered.  Just do the ForkCDR on the polarity reversal.  This will give you
> two CDR events for a good call and one for a bad call.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha
> Sent: Friday, October 02, 2009 2:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] TDM410P - False Answer Supervision
>
> Anyone else having this issue with TDM410P card or anyone with a solution?
>
> Please advise... Thanks
>
> Cheers,
> Nitesh
>
>
>
> Martin wrote:
>   
>> Are you in US ?
>>
>> do you have the proper keywords in zapata.conf/chan_dahdi.conf like
>> callprogress=yes etc ?
>>
>> Martin
>>
>> On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha 
>> 
> wrote:
>   
>>   
>> 
>>> Danny,
>>>
>>> Thanks for your reply...
>>>
>>> Yes these are POTS line and I am not calling myself... Any other
>>> suggestions?
>>>
>>> Cheers,
>>> Nitesh
>>>
>>>
>>> Danny Nicholas wrote:
>>> 
>>>   
>>>> Assuming you're using POTS, you probably won't have much luck with this.
>>>> 
> If
>   
>>>> you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and
>>>> 
> asterisk
>   
>>>> won't process the line until you pick up and punch a dtmf key.  If you
>>>> 
> are
>   
>>>> using E1 or PRI, there is more hope for you.
>>>>
>>>> -Original Message-
>>>> From: asterisk-users-boun...@lists.digium.com
>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh
>>>> 
> Divecha
>   
>>>> Sent: Thursday, October 01, 2009 4:42 PM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Subject: [asterisk-users] TDM410P - False Answer Supervision
>>>>
>>>> Hello All,
>>>>
>>>> Can anyone help me with False Answer Supervision problem with TDM410P
>>>> card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and
>>>> everything works fine except the Answer supervision...
>>>>
>>>> When the call hits Asterisk it sends the call to one of the TDM410 card
>>>> and the call is answered immediately while the call is still in
>>>> progress... Here is the debug output: -
>>>>
>>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing
>>>> '3602045'
>>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring
>>>> dialing...
>>>> -- Called G2/3602045
>>>> [Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>>> deferred digit string: T3602045w
>>>> [Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>>> dialing, but waiting for progress detection before doing more...
>>>> -- DAHDI/8-1 answered SIP/9223421808-091b3f50
>>>> -- Hungup 'DAHDI/8-1'
>>>> =
>>>>
>>>> The connect message is sent back immediately when " DAHDI/8-1 answered
>>>> SIP/9223421808-091b3f50" while the call is still in progress... If the
>>>> call is hang up without answer the sender gets Normal Code 16 while it
>>>> suppose to be "Abandoned Call".
>>>>
>>>>
>>>>
>>>> The Polarity Reversal only works when call is ANSWERED... Here is the
>>>> debug log: -
>>>>
>>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing
>>>> '3312808'
>>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi

Re: [asterisk-users] TDM410P - False Answer Supervision

[Nitesh Divecha]

Anyone else having this issue with TDM410P card or anyone with a solution?

Please advise... Thanks

Cheers,
Nitesh



Martin wrote:
> Are you in US ?
>
> do you have the proper keywords in zapata.conf/chan_dahdi.conf like
> callprogress=yes etc ?
>
> Martin
>
> On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha  
> wrote:
>   
>> Danny,
>>
>> Thanks for your reply...
>>
>> Yes these are POTS line and I am not calling myself... Any other
>> suggestions?
>>
>> Cheers,
>> Nitesh
>>
>>
>> Danny Nicholas wrote:
>> 
>>> Assuming you're using POTS, you probably won't have much luck with this.  If
>>> you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk
>>> won't process the line until you pick up and punch a dtmf key.  If you are
>>> using E1 or PRI, there is more hope for you.
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha
>>> Sent: Thursday, October 01, 2009 4:42 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [asterisk-users] TDM410P - False Answer Supervision
>>>
>>> Hello All,
>>>
>>> Can anyone help me with False Answer Supervision problem with TDM410P
>>> card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and
>>> everything works fine except the Answer supervision...
>>>
>>> When the call hits Asterisk it sends the call to one of the TDM410 card
>>> and the call is answered immediately while the call is still in
>>> progress... Here is the debug output: -
>>>
>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing
>>> '3602045'
>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring
>>> dialing...
>>> -- Called G2/3602045
>>> [Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>> deferred digit string: T3602045w
>>> [Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>> dialing, but waiting for progress detection before doing more...
>>> -- DAHDI/8-1 answered SIP/9223421808-091b3f50
>>> -- Hungup 'DAHDI/8-1'
>>> =
>>>
>>> The connect message is sent back immediately when " DAHDI/8-1 answered
>>> SIP/9223421808-091b3f50" while the call is still in progress... If the
>>> call is hang up without answer the sender gets Normal Code 16 while it
>>> suppose to be "Abandoned Call".
>>>
>>>
>>>
>>> The Polarity Reversal only works when call is ANSWERED... Here is the
>>> debug log: -
>>>
>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing
>>> '3312808'
>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring
>>> dialing...
>>> -- Called G2/3312808
>>> [Oct  2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>> deferred digit string: T3312808w
>>> [Oct  2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>> dialing, but waiting for progress detection before doing more...
>>> -- DAHDI/8-1 answered SIP/9765782184-091b9678
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event:
>>> Ignore switch to REVERSED Polarity on channel 8, state 6
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event:
>>> Ignoring Polarity switch to IDLE on channel 8, state 6
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event:
>>> Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0,
>>> aonp= 1, honp= 0, pdelay= 600, tv= 301564043
>>> -- Hungup 'DAHDI/8-1'
>>> =
>>>
>>> Please help...
>>>
>>> Cheers,
>>> Nitesh
>>>
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>> Register Now: http://www.astricon.net
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Re: [asterisk-users] TDM410P - False Answer Supervision

[Nitesh Divecha]

Thanks Martin,

Well the Asterisk is in Fiji and we have check with the Telco on 
"Reverse Polarity" and they said it is setup...

Here is my chan_dahdi.conf:-

#include dahdi-channels.conf

[channels]
language=en
context=incoming
signalling=fxs_ks
busydetect=yes
callprogress=yes
usecallerid=yes
;hanguponpolarityswitch=yes
answeronpolarityswitch=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=2.0
txgain=3.0
group=1
callgroup=1
pickupgroup=1

channel => 1-4
channel => 5-8


Cheers,
Nitesh





Martin wrote:
> Are you in US ?
>
> do you have the proper keywords in zapata.conf/chan_dahdi.conf like
> callprogress=yes etc ?
>
> Martin
>
> On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha  
> wrote:
>   
>> Danny,
>>
>> Thanks for your reply...
>>
>> Yes these are POTS line and I am not calling myself... Any other
>> suggestions?
>>
>> Cheers,
>> Nitesh
>>
>>
>> Danny Nicholas wrote:
>> 
>>> Assuming you're using POTS, you probably won't have much luck with this.  If
>>> you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk
>>> won't process the line until you pick up and punch a dtmf key.  If you are
>>> using E1 or PRI, there is more hope for you.
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha
>>> Sent: Thursday, October 01, 2009 4:42 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [asterisk-users] TDM410P - False Answer Supervision
>>>
>>> Hello All,
>>>
>>> Can anyone help me with False Answer Supervision problem with TDM410P
>>> card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and
>>> everything works fine except the Answer supervision...
>>>
>>> When the call hits Asterisk it sends the call to one of the TDM410 card
>>> and the call is answered immediately while the call is still in
>>> progress... Here is the debug output: -
>>>
>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing
>>> '3602045'
>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring
>>> dialing...
>>> -- Called G2/3602045
>>> [Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>> deferred digit string: T3602045w
>>> [Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>> dialing, but waiting for progress detection before doing more...
>>> -- DAHDI/8-1 answered SIP/9223421808-091b3f50
>>> -- Hungup 'DAHDI/8-1'
>>> =
>>>
>>> The connect message is sent back immediately when " DAHDI/8-1 answered
>>> SIP/9223421808-091b3f50" while the call is still in progress... If the
>>> call is hang up without answer the sender gets Normal Code 16 while it
>>> suppose to be "Abandoned Call".
>>>
>>>
>>>
>>> The Polarity Reversal only works when call is ANSWERED... Here is the
>>> debug log: -
>>>
>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing
>>> '3312808'
>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring
>>> dialing...
>>> -- Called G2/3312808
>>> [Oct  2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>> deferred digit string: T3312808w
>>> [Oct  2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>> dialing, but waiting for progress detection before doing more...
>>> -- DAHDI/8-1 answered SIP/9765782184-091b9678
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event:
>>> Ignore switch to REVERSED Polarity on channel 8, state 6
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event:
>>> Ignoring Polarity switch to IDLE on channel 8, state 6
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event:
>>> Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0,
>>> aonp= 1, honp= 0, pdelay= 600, tv= 301564043
>>> -- Hungup 'DAHDI/8-1'
>>> =
>>>
>>> Please help...
>>>
>>> Cheers,
>>> Nitesh
>>>
>>

Re: [asterisk-users] TDM410P - False Answer Supervision

[Nitesh Divecha]

Danny,

Thanks for your reply...

Yes these are POTS line and I am not calling myself... Any other 
suggestions?

Cheers,
Nitesh


Danny Nicholas wrote:
> Assuming you're using POTS, you probably won't have much luck with this.  If
> you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk
> won't process the line until you pick up and punch a dtmf key.  If you are
> using E1 or PRI, there is more hope for you.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha
> Sent: Thursday, October 01, 2009 4:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] TDM410P - False Answer Supervision
>
> Hello All,
>
> Can anyone help me with False Answer Supervision problem with TDM410P 
> card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and 
> everything works fine except the Answer supervision...
>
> When the call hits Asterisk it sends the call to one of the TDM410 card 
> and the call is answered immediately while the call is still in 
> progress... Here is the debug output: -
>
> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing 
> '3602045'
> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring 
> dialing...
> -- Called G2/3602045
> [Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent 
> deferred digit string: T3602045w
> [Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done 
> dialing, but waiting for progress detection before doing more...
> -- DAHDI/8-1 answered SIP/9223421808-091b3f50
> -- Hungup 'DAHDI/8-1'
> =
>
> The connect message is sent back immediately when " DAHDI/8-1 answered 
> SIP/9223421808-091b3f50" while the call is still in progress... If the 
> call is hang up without answer the sender gets Normal Code 16 while it 
> suppose to be "Abandoned Call".
>
>
>
> The Polarity Reversal only works when call is ANSWERED... Here is the 
> debug log: -
>
> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing 
> '3312808'
> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring 
> dialing...
> -- Called G2/3312808
> [Oct  2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent 
> deferred digit string: T3312808w
> [Oct  2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done 
> dialing, but waiting for progress detection before doing more...
> -- DAHDI/8-1 answered SIP/9765782184-091b9678
> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event: 
> Ignore switch to REVERSED Polarity on channel 8, state 6
> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event: 
> Ignoring Polarity switch to IDLE on channel 8, state 6
> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event: 
> Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0, 
> aonp= 1, honp= 0, pdelay= 600, tv= 301564043
> -- Hungup 'DAHDI/8-1'
> =
>
> Please help...
>
> Cheers,
> Nitesh
>
>
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[asterisk-users] TDM410P - False Answer Supervision

[Nitesh Divecha]

Hello All,

Can anyone help me with False Answer Supervision problem with TDM410P 
card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and 
everything works fine except the Answer supervision...

When the call hits Asterisk it sends the call to one of the TDM410 card 
and the call is answered immediately while the call is still in 
progress... Here is the debug output: -

[Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing 
'3602045'
[Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring 
dialing...
-- Called G2/3602045
[Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent 
deferred digit string: T3602045w
[Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done 
dialing, but waiting for progress detection before doing more...
-- DAHDI/8-1 answered SIP/9223421808-091b3f50
-- Hungup 'DAHDI/8-1'
=

The connect message is sent back immediately when " DAHDI/8-1 answered 
SIP/9223421808-091b3f50" while the call is still in progress... If the 
call is hang up without answer the sender gets Normal Code 16 while it 
suppose to be "Abandoned Call".



The Polarity Reversal only works when call is ANSWERED... Here is the 
debug log: -

[Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing 
'3312808'
[Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring 
dialing...
-- Called G2/3312808
[Oct  2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent 
deferred digit string: T3312808w
[Oct  2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done 
dialing, but waiting for progress detection before doing more...
-- DAHDI/8-1 answered SIP/9765782184-091b9678
[Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event: 
Ignore switch to REVERSED Polarity on channel 8, state 6
[Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event: 
Ignoring Polarity switch to IDLE on channel 8, state 6
[Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event: 
Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0, 
aonp= 1, honp= 0, pdelay= 600, tv= 301564043
-- Hungup 'DAHDI/8-1'
=

Please help...

Cheers,
Nitesh


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[asterisk-users] ISDN Error Code 42

[Nitesh Divecha]

Hello All,

Just got one general question on ISDN error code 42. As per Cisco docs 
and Wiki 
(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause) it 
says "Switch equipment Congestion" with explanation "The destination 
cannot be reached because the network switching equipment is temporarily 
overloaded"...

My question is are the calls treated as "Failed" and I will be pulled 
away from the routing? Is this similar to ISDN error code 34's?

Please help...

Cheers,
Nitesh


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[asterisk-users] Asterisk/GXW410x IP Analog Gateway

[Nitesh Divecha]

Hello All,

I am trying to setup a small system where Nextone Softswitch will send 
traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog 
Gateway but for some odd reasons the call are flashed back from 
Grandstream to Asterisk and creating a Black loop...

I did follow the instructions provided by Grandstream support but it 
doesn't seems to be working... 
http://www.grandstream.com/documents/GXW410xwithAsteriskConfiguration.pdf

OS: Ubuntu 8
Asterisk 1.4.22.1

Anyone implemented Grandstream GXW410x IP Gateway with Asterisk and can 
share config?

Cheers,
Nitesh


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[asterisk-users] Error while Compiling zaptel-1.4.11

[Nitesh Divecha]

Hello All,

This is my third freshly installed and updated CentOS 5.1 with installed 
Digium 4-port Analog card and while compiling Zaptel I am getting this 
error. If I run "./install_preq test" and "./install_preq install" it 
says "Install Successfully".

Error
=

CC [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/../voicebus.o
LD [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/wcte12xp.o
CC [M] /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o
In file included from /usr/src/zaptel-1.4.11/kernel/xpp/xpd.h:26,
from /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.c:27:
/usr/src/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types 
for ‘bool’
include/linux/types.h:36: error: previous declaration of ‘bool’ was here
make[4]: *** [/usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1
make[3]: *** [/usr/src/zaptel-1.4.11/kernel/xpp] Error 2
make[2]: *** [_module_/usr/src/zaptel-1.4.11/kernel] Error 2
make[2]: Leaving directory `/usr/src/kernels/2.6.18-92.1.6.el5-i686'
make[1]: *** [modules] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4.11'
make: *** [all] Error 2
[EMAIL PROTECTED] zaptel-1.4.11]#


Can anyone help...

Cheers,
Nitesh



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[asterisk-users] MOH and Licensed G729 codec

[Nitesh Divecha]

Hello All,

Recently, I build three Asterisk 1.4 box and installed licensed copy of 
G729 codec. Before installing the G729 codec I tested the MOH on all 
three Asterisks box and it was working fine. So I install G729 codec and 
retested MOH and it was all wavy... Meaning the music was going up and 
down and missing bits and pieces and choppy...

Any idea what did I do wrong? The MOH files are the default ones which 
comes with Asterisk.

Cheers,
Neel


 

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Re: [asterisk-users] AGI / Voicemail Que

[Nitesh Divecha]

Thanks Doug,

I tried that but it didn't work either... As per Wiki 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail it 
has a statement that starting from 1.4-trunk FLAG must be pass using a 
pipe sign '|'.

I have other Asterisk 1.2 running with FreePBX and I went over the agi 
code, I saw its passing FLAG using pipe sign '|'.

So now I am kinda confused...

Cheers,
Nitesh





Doug Lytle wrote:
> Nitesh Divecha wrote:
>   
>> ([EMAIL PROTECTED]|b)
>>
>> Any suggestions... By the way I am running Asterisk 1.2.18
>>
>>   
>> 
>
> I believe under 1.2.x it would be [EMAIL PROTECTED]
>
> One of my older dial plans lists:
>
> s-BUSY,1,Voicemail([EMAIL PROTECTED])
>
> Doug
>
>
>   


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[asterisk-users] AGI / Voicemail Que

[Nitesh Divecha]

Hello All,

I have my own AGI script running and I am trying to push the call to 
voice mail when Busy, Unavailable and Not Answered.

Everything is working fine but the only problem is voice mail greetings 
for Busy and Unavailable is not played. By default only "Temp Greetings" 
voice mail greetings is played. I am passing the correct parameters for 
Busy => 'b', Unavailable => 'u' and default goes to "Not Answered".

Here is a code sample: -

exec("VOICEMAIL",  $arr);
} elseif ($status == "CHANUNAVAIL"){
$arr = array("[EMAIL PROTECTED]", 'u');
$agi->exec("VOICEMAIL",  $arr);
} else {
$arr = array("[EMAIL PROTECTED]");
$agi->exec("VOICEMAIL",  $arr);
}
?>

Here is the AGI Debug message: -
-- AGI Script Executing Application: (VOICEMAIL) Options: 
([EMAIL PROTECTED]|b)
-- Playing '/var/spool/asterisk/voicemail/default/2481237766/temp' 
(language 'en')

As you can see I am passing the correct parameter for BUSY => |b, but 
Asterisk is only playing the "temporary greetings".

Any suggestions... By the way I am running Asterisk 1.2.18

Cheers,
Nitesh



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Re: [asterisk-users] SAY TIME + PHPAGI + Timezone

[Nitesh Divecha]

Thanks everyone for the feedback... Manage to prompt time using EXEC 
with the SayUnixTime.

Here is the snapshot of the timezone: -

// Get current time
$currentTime = time();

// Set the offset
$offset = 3;

// Modified time
$modifiedTime = $currentTime + ($offset * 60 * 60);
debug("Current time: $currentTime", 3);
debug("Offset time: $modifiedTime", 3);

// Say unix time
$agi->exec("SayUnixTime", "$modifiedTime,EST5EDT,ABdY \'digits/at\' IMp");

Cheers,
Nitesh




Tilghman Lesher wrote:
> On Friday 18 January 2008 14:13:55 Nitesh Divecha wrote:
>   
>> Is there any way to change the timezone on the fly? I have this little
>> time clock program running on Asterisk system developed using PHPAGI.
>> Currently, whenever user logs in, Asterisk will prompt the current
>> system time using "$agi->say_time();" which executes "SAY TIME". Now the
>> current timezone set on the system is "PST", and I have a request to
>> prompt multiple timezones based on the users location.
>> 
>
> Don't use SAY TIME.  Use EXEC with the SayUnixTime application and the
> appropriate arguments.
>
>   


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[asterisk-users] SAY TIME + PHPAGI + Timezone

[Nitesh Divecha]

Hello All,

Is there any way to change the timezone on the fly? I have this little 
time clock program running on Asterisk system developed using PHPAGI. 
Currently, whenever user logs in, Asterisk will prompt the current 
system time using "$agi->say_time();" which executes "SAY TIME". Now the 
current timezone set on the system is "PST", and I have a request to 
prompt multiple timezones based on the users location.

First part is easy to lookup from which area code the user is calling, 
now the second part is to set the timezone based on the area code and 
prompt the users correct time.

For example, if 248 (MI) user dials into the system, then time clock has 
to prompt EST time and if 714 (CA) user dials in then prompt PST time.

Any suggestions... Thanks

Cheers,
Nitesh




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[asterisk-users] Maximum retries/no reply to our critical packet

[Nitesh Divecha]

Hello All,

Got one customer and he is getting disconnection within 15 seconds when 
he tries to make outbound calls. Initially, it was working fine without 
any glitches... Other customers on the same system are working fine, its 
just with this customer only.

This is the error message thrown by Asterisk on the CLI: -

Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1228 retrans_pkt: Maximum 
retries exceeded on transmission [EMAIL PROTECTED] for seqno 
102 (Critical Response)
Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1245 retrans_pkt: Hanging up 
call [EMAIL PROTECTED] - no reply to our critical packet.

Customer can receive inbound calls without any disconnections, its just 
when he tries to make outbound calls.

All outbound calls are sent to Nextone SoftSwitch and default codec is 
G729a. Customer has Linksys SPA-2102 - firmware ver 3.3.6 and Asterisk 
version 1.2.18.

Thanking in advance...

Cheers,
Nitesh



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Re: [asterisk-users] Injecting a sound file into a bridged call

[Nitesh Divecha]

Hello All,

Same here need for such functionality but for my application instead of 
playing sound file, I need ivr menu selection... So for example, if 
manager and employee are in call there should be ivr menu selection 
playing (Press 1 if you are coming late, Press 2 if you are sick...) and 
manager can select any of these options while employee is listening to 
it...

Cheers,
Nitesh



Atis Lezdins wrote:
> On 08/10/2007, Girts Graudins <[EMAIL PROTECTED]> wrote:
>   
>>> I'm looking for a way to play a sound file to an already established
>>> bridged call.  It is meant for one party, but it's ok if both parties
>>> would hear it.  Ideally, I'd like to be able to trigger this from the
>>> Management Interface with something like:
>>>   
>
> I'm also in need for such functionality, the only difference is that i need 
> for both channels to hear the message. As i have read press releases, there 
> will be something similar available in 1.6. If you succeed, please give us a 
> note - how it can be done.
>
>   
>>> 2)  I've seen "whisper"-type of functionality associated with meetme
>>> rooms, but I'd rather not set up a dynamic meetme room for each call I'm
>>> bridging;
>>>   
>
> Well, you can create conference dynamically whenever you need to play the 
> file. I started working on this, and have found several bugs regarding this, 
> but they should be fixed in 1.4.12
>
> Idea is to Redirect() trough AMI both channels to dynamical conference, and 
> then attach call with Playback() to the same conference. For now, the 
> Redirect() part is working fine, but due to lack of time, i haven't got 
> further.
>
> On Monday 08 October 2007 14:13:38 Jaswinder Singh wrote:
>   
>> See chanspy in asterisk 1.4 , it also has a whisper mode and you can talk
>> to one party http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy . But
>> i dont know  how to play  a recorded file in it .
>> 
>
> My collegues tried this but unsuccessfully. The basic idea is to use local 
> channels - one is bridged to Chanspy() and second to Playback(). I'm not sure 
> what is the problem, but theoretically also this should work.
>
> Regards,
> Atis
>
>   


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[asterisk-users] Asterisk Keep Loosing Registration

[Nitesh Divecha]

Hello All,

For some odd reasons my Asterisk is keep on loosing registration of my 
SIP devices. On the SIP device it shows I am RESISTED but when I do "sip 
show peers" it shows my sip endpoints are "UNREACHABLE". And it keeps on 
flapping "Peer '903456' is now UNREACHABLE!" and "Peer '903456' 
is now REACHABLE!"...

I changed my maxexpiry and defaultexpiry to 3600 in sip.conf but still 
it didn't help.

I am using Asterisk 1.2.18 with Real-Time config.

Any help will be appreciated...

Cheers,
Nitesh




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[asterisk-users] DTMF Problem with International Calls

[Nitesh Divecha]

Hello All,

Does anyone knows a good carrier who can pass DTMF tone while doing Call 
Back? Currently, the Call Back system works within US, but as soon as 
international users tries to enter phone number the system does not 
understand the tones.

I tried to change the sip config to inband, auto, RFC2833 but it didnt 
work... So I suspect its my VoIP Carrier who doesn't pass the 
International DTMF tones.

Any suggestions?

Cheers,
Nitesh


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[asterisk-users] Asterisk Died message

[Nitesh Divecha]

Hello All,

Anyone knows what does this error message means and where to check for 
the cause and why it happened?

"Asterisk on hyperion exited on signal 11. Might want to take a peek."

But when I check Asterisk, its running fine...

Cheers,
Nitesh



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Re: [asterisk-users] Monitor System using AGI Scripts

[Nitesh Divecha]

Thanks man,

That's what I was looking right now... to use Nagios with asterisk plug-ins.

Cheers,
Nitesh
 

James FitzGibbon wrote:
> On 8/29/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>> wrote:
>
>
> Basically, it would be a totally different system running Asterisk
> with
> AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not
> specifically monitoring ports (80, 21, 25) but whole system. If system
> timeouts then AGI scripts are triggered and notify system admin.
>
>
> You'd be better to monitor using something like Nagios or one of the 
> other open-source monitoring systems, then have the notification 
> script (which should be customizable in your monitoring system) write 
> a .call file to make Asterisk dial out and tell the sysadmin.
>
> To use the Asterisk dialplan to schedule and cycle checks of services 
> . erm. no.
>
> -- 
> j.
> 
>
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Re: [asterisk-users] Monitor System using AGI Scripts

[Nitesh Divecha]

Thanks Jared,

Basically, it would be a totally different system running Asterisk with 
AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not 
specifically monitoring ports (80, 21, 25) but whole system. If system 
timeouts then AGI scripts are triggered and notify system admin.

I saw one PHP-AGI example "ping.php", might able to modify abit and work 
around... but to ping a systems 24/7 its chaos...

Cheers,
Nitesh



 
Jared Smith wrote:
> On Wed, 2007-08-29 at 10:46 -0400, Nitesh Divecha wrote:
>   
>> Anyone using AGI scripts to monitor their systems?
>>
>> Something like if the system goes down, AGI script will be triggered and 
>> system admin will be notified saying "System XYZ has gone down"...
>> 
>
> If the system goes down, how would an AGI script get triggered?  I know
> lots of people using the Asterisk Manager Interface to monitor their
> Asterisk systems, or res_snmp on Asterisk 1.4.  You'd probably be better
> suited to look at those first.
>
>
>   


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[asterisk-users] Monitor System using AGI Scripts

[Nitesh Divecha]

Hello All,

Anyone using AGI scripts to monitor their systems?

Something like if the system goes down, AGI script will be triggered and 
system admin will be notified saying "System XYZ has gone down"...

Any suggestions...

Cheers,
Nitesh


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Hello All,

Stable release of A2Billing has solved most of my problems and so far 
everything is OK...

Right now the only problem I am facing with my SIP clients are: -
- Three-way Calling
   Three-way calling works fine, but when SIP client hangs up the 
call, the other two channels are still active and talking.

- Call Forwarding
   I created the context for *72, *73, *90, *91, *52, and *53. SIP 
client can enable and disable but it never works because "a2billing.php" 
will time out and hang up the SIP channel.

- Voice Mail
   I created the context for voice mail, but the calls will never go 
to voice mail because "a2billing.php" after 60 sec will hang up the 
channel.

No doubt A2Billing is a great software, but the above features are also 
essential for home SIP users...

Anyone can show or share their setup if they have implemented the above 
features with A2Billing Software.

Cheers,
Nitesh



Al Bochter wrote:
> In a2billing just change the 9 to what you need it is right in the 
> conf file.
> Best regards,
>
> Al Bochter
> Bochter Services
>
> --
> Need to call me use our web phone at the link below
> http://www.bochterservices.com/voip/iaxphone.php?cn=250
> --
> Can you WIN gold today? Click on the link and see.
> http://www.bochterservices.com/?t=USbill_email
> --
> Need cash we buy silver and gold
> ----------
>
>
> Nitesh Divecha wrote:
>> Thanks everyone for the input...
>>
>> In real world we can not ask the customers to dial 9, if they want to 
>> call another SIP user... and trust me its confusing for a customer 
>> also... meaning when to dial 9 and when to not...
>>
>> We have a custom proprietary system which does this part very well... 
>> Before it sends the call on a Trunk it will check the DID, if it exists 
>> within the local system. If it does then it will just use IP to IP call, 
>> else send the call to Trunk...
>>
>> I think its possible to do this by creating some basic dial plans... 
>> Same like creating local extensions.
>>
>> Cheers,
>> Nitesh
>>
>>
>>
>>
>> John Novack wrote:
>>   
>>> Given that Asterisk is modeled on, in the telephone industry, an 
>>> obsolete PBX design, without many of the modern day hybrid features, and 
>>> only recently has any effort been made to provide buttons and lights for 
>>> "lines" ( Is that yet working in 1.4??) one would have to do some very 
>>> careful number parsing to not use a trunk digit.
>>>
>>> If every phone in the system had buttons and lights representing 
>>> external connections and internal connections on other button(s) ( 
>>> intercom ) this wouldn't be an issue.
>>> Most "legacy" systems have been able to do this for the last 20 years or so.
>>>
>>> John Novack
>>>
>>>
>>> Nitesh Divecha wrote:
>>>   
>>> 
>>>> Thanks man,
>>>>
>>>> Is there any other way without dialing 9... it will be kinda pain for a 
>>>> customer to dial 9 every time and plus they need to know also...
>>>>
>>>> Is there any intelligent way to identify? if its a local SIP then don't 
>>>> route to Trunk else route to Trunk.
>>>>
>>>> Cheers,
>>>> Nitesh
>>>>
>>>>
>>>> Guillermo Salas M. wrote:
>>>>   
>>>> 
>>>>   
>>>>> On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote:
>>>>>   
>>>>> 
>>>>>   
>>>>> 
>>>>>> Thanks man...
>>>>>>
>>>>>> So far everything worked as expected...
>>>>>>
>>>>>> How can I make internal calls stay within the PBX. For example, when
>>>>>> one 
>>>>>> SIP-Friend tries to call another SIP-Friend without sending the call
>>>>>> out 
>>>>>> on Trunk and receive it back. Same like dialing from one extension 
>>>>>> number to another extension.
>>>>>>
>>>>>> My SIP-Friends are using US DID numbers and I would like to keep the 
>>>>>> local calls within the network.
>>>>>>
>>>>>> Right now when I try to call other SIP-Friend, I get 

[asterisk-users] SET EXTENSION

[Nitesh Divecha]

Hello All,

How can I SET EXTENSION from context?

This is my context: -

[docall-usa]
exten => _NXXNXX,1,Answer
exten => _NXXNXX,n,Set() ; <>
exten => _NXXNXX,n,DeadAGI(dousacall.php|1)
exten => _NXXNXX,n,Hangup

I need to add 1 in front of ${EXTEN} and then send the call to dousa.php.

Set(CALLERID(number)=1${EXTEN}) will set the callerID to that 
extension... But I want to add '1' to my extension.

Can anyone please put some light... what I am missing here...

Cheers,
Nitesh



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[asterisk-users] Increase Volume on AGI

[Nitesh Divecha]

Hello All,

I have my AGI scripts streaming wav files and I would like to increase 
the volume on it.
Is there any way to increase the volume on outbound SIP trunks, instead 
of me changing the wav file one by one?

Please help?

Cheers,
Nitesh


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[asterisk-users] Callback DTMF Problem

[Nitesh Divecha]

Hello All,

I don't understand where is the problem...
I have Callback setup and it works fine when tested within US. Works 
fine meaning the DTMF tones are passed when prompted to enter the phone 
number. But when I test with some international countries, callback 
works but DTMF tones are not passed...

Is it: -
a) Asterisk problem?
b) Callback problem?
c) VoIP provider problem?


Under sip.conf I have "dtmfmode = auto".

Please help...

Cheers,
Nitesh



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Re: [asterisk-users] LumenVox Speech Recognition

[Nitesh Divecha]

Dean,

Can the LumenVox Speech Recognition engine understand numbers?
Sorry to ask stupid questions but kinda curious... as for my application 
all I want is to the software to understand the numbers and provide me 
the output.

Cheers,
Nitesh


Dean Collins wrote:
> No they have a standalone solution - lol NLVR "is" a whole separate
> server (or server farm) in most onsite installations.
>
>
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +61-2-9016-5642 (Sydney in-dial).
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of mitcheloc
>> Sent: Saturday, 11 August 2007 7:29 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] LumenVox Speech Recognition
>>
>> Dean,
>>
>> Hmm.. I was hoping something that could be used with Asterisk on the
>> machine locally... Nuance doesn't seem to offer that.
>>
>> On 8/11/07, Dean Collins <[EMAIL PROTECTED]> wrote:
>> 
>>> Nuance etc.  and Steve to answer your questions - lumenvox just
>>>   
> doesn't
>   
>>> have the engine or phonetic capabilities that some of the the larger
>>> systems have.
>>>
>>> Like I said before - I've been stunned considering how cheap it is
>>>   
> how
>   
>>> good it is but. if you are looking for a less defined utterance
>>> structure it has limitations.
>>>
>>>
>>>
>>> Regards,
>>>
>>> Dean Collins
>>> Cognation Pty Ltd
>>> [EMAIL PROTECTED]
>>> +1-212-203-4357 Ph
>>> +61-2-9016-5642 (Sydney in-dial).
>>>
>>>
>>>   
>>>> -Original Message-
>>>> From: [EMAIL PROTECTED]
>>>> 
> [mailto:asterisk-users-
>   
>>>> [EMAIL PROTECTED] On Behalf Of mitcheloc
>>>> Sent: Saturday, 11 August 2007 3:25 PM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Subject: Re: [asterisk-users] LumenVox Speech Recognition
>>>>
>>>> Dean,
>>>>
>>>> Are you aware of any better options for speech recognition?
>>>> 
> (though
>   
>>>> I'm sure more expensive)
>>>>
>>>> On 8/11/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:
>>>>     
>>>>> Thanks Dean... will update you on the progress...
>>>>>
>>>>> Cheers,
>>>>> Nitesh
>>>>>
>>>>>
>>>>>
>>>>> Dean Collins wrote:
>>>>>   
>>>>>> Hi Nitesh - yep great place to start.
>>>>>>
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> Dean Collins
>>>>>> Cognation Pty Ltd
>>>>>> [EMAIL PROTECTED]
>>>>>> +1-212-203-4357 Ph
>>>>>> +61-2-9016-5642 (Sydney in-dial).
>>>>>>
>>>>>>
>>>>>>
>>>>>> 
>>>>>>> -Original Message-
>>>>>>> From: [EMAIL PROTECTED]
>>>>>>>   
>>> [mailto:asterisk-users-
>>>   
>>>>>>> [EMAIL PROTECTED] On Behalf Of Nitesh Divecha
>>>>>>> Sent: Saturday, 11 August 2007 11:40 AM
>>>>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>>>>> Subject: Re: [asterisk-users] LumenVox Speech Recognition
>>>>>>>
>>>>>>> Thanks Dean and Steve,
>>>>>>>
>>>>>>> I am planning to use for my IVR notification application
>>>>>>>   
> which is
>   
>>>>>> built
>>>>>>
>>>>>> 
>>>>>>> using PHPAGI and A2Billing (Callback, Calling Card).
>>>>>>> I saw the $50.00 Starter kit does it provide some
>>>>>>>   
> functionality?
>   
>>>>>>> Cheers,
>>>>>>> Nitesh
>>>>>>>
>>>>>>>
>>>>>>> Dean Collins wrote:
>>>>>>>
>>>>>>>   
>>>>>>>> Hi Steve, no I'm no expert at all I do however (or did)
>>

Re: [asterisk-users] LumenVox Speech Recognition

[Nitesh Divecha]

Thanks Dean... will update you on the progress...

Cheers,
Nitesh



Dean Collins wrote:
> Hi Nitesh - yep great place to start.
>
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +61-2-9016-5642 (Sydney in-dial).
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Nitesh Divecha
>> Sent: Saturday, 11 August 2007 11:40 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] LumenVox Speech Recognition
>>
>> Thanks Dean and Steve,
>>
>> I am planning to use for my IVR notification application which is
>> 
> built
>   
>> using PHPAGI and A2Billing (Callback, Calling Card).
>> I saw the $50.00 Starter kit does it provide some functionality?
>>
>> Cheers,
>> Nitesh
>>
>>
>> Dean Collins wrote:
>> 
>>> Hi Steve, no I'm no expert at all I do however (or did) have an
>>> interest in building a far more comprehensive solution for an ASP
>>> solution combining other solutions that would have helped the
>>>   
> asterisk
>   
>>> community however could never get it off the ground.
>>>
>>> Nitesh to answer your original question...Lumenvox is great value
>>>   
> for
>   
>>> the money and works well - however there are limitations but for 90%
>>>   
> of
>   
>>> applications will work great.
>>>
>>>
>>>
>>>
>>> Regards,
>>>
>>> Dean Collins
>>> Cognation Pty Ltd
>>> [EMAIL PROTECTED]
>>> +1-212-203-4357 Ph
>>> +61-2-9016-5642 (Sydney in-dial).
>>>
>>>
>>>
>>>   
>>>> -----Original Message-
>>>> From: [EMAIL PROTECTED]
>>>> 
> [mailto:asterisk-users-
>   
>>>> [EMAIL PROTECTED] On Behalf Of Steve Totaro
>>>> Sent: Saturday, 11 August 2007 10:55 AM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Subject: Re: [asterisk-users] LumenVox Speech Recognition
>>>>
>>>> Dean Collins is probably the list expert on this.
>>>>
>>>> Thanks,
>>>> Steve Totaro
>>>>
>>>> Nitesh Divecha wrote:
>>>>
>>>> 
>>>>> Hello All,
>>>>>
>>>>> While looking for solution to solve my Callback DTMF problem, I
>>>>>   
> came
>   
>>>>> across LumenVox Speech Recognition software.
>>>>>
>>>>> Has anyone tried out? Need some feedback before I purchase it...
>>>>>
>>>>>   
>>> Please
>>>
>>>   
>>>>> help...
>>>>>
>>>>> Cheers,
>>>>> Nitesh
>>>>>
>>>>>
>>>>>
>>>>> ___
>>>>> --Bandwidth and Colocation Provided by
>>>>>   
> http://www.api-digital.com--
>   
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>>>>>
>>>>>
>>>>>
>>>>>   
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Re: [asterisk-users] LumenVox Speech Recognition

[Nitesh Divecha]

Thanks Dean and Steve,

I am planning to use for my IVR notification application which is built 
using PHPAGI and A2Billing (Callback, Calling Card).
I saw the $50.00 Starter kit does it provide some functionality?

Cheers,
Nitesh


Dean Collins wrote:
> Hi Steve, no I'm no expert at all I do however (or did) have an
> interest in building a far more comprehensive solution for an ASP
> solution combining other solutions that would have helped the asterisk
> community however could never get it off the ground.
>
> Nitesh to answer your original question...Lumenvox is great value for
> the money and works well - however there are limitations but for 90% of
> applications will work great.
>
>
>
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +61-2-9016-5642 (Sydney in-dial).
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Steve Totaro
>> Sent: Saturday, 11 August 2007 10:55 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] LumenVox Speech Recognition
>>
>> Dean Collins is probably the list expert on this.
>>
>> Thanks,
>> Steve Totaro
>>
>> Nitesh Divecha wrote:
>> 
>>> Hello All,
>>>
>>> While looking for solution to solve my Callback DTMF problem, I came
>>> across LumenVox Speech Recognition software.
>>>
>>> Has anyone tried out? Need some feedback before I purchase it...
>>>   
> Please
>   
>>> help...
>>>
>>> Cheers,
>>> Nitesh
>>>
>>>
>>>
>>> ___
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>>   
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[asterisk-users] LumenVox Speech Recognition

[Nitesh Divecha]

Hello All,

While looking for solution to solve my Callback DTMF problem, I came 
across LumenVox Speech Recognition software.

Has anyone tried out? Need some feedback before I purchase it... Please 
help...

Cheers,
Nitesh



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[asterisk-users] AGI SAY TIME

[Nitesh Divecha]

Hello all,

Can anyone help me with SAY TIME.
Every time I ask to say time, it gives me wrong time.
I want the system to say time, what ever I give to say.
Is it possible?

Cheers,
Nitesh



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Re: [asterisk-users] .call file problem

[Nitesh Divecha]

Thanks Eric,

It solved the problem by having a blank line... I wonder why we need a 
blank line...

Cheers,
Nitesh



Eric "ManxPower" Wieling wrote:
> Make sure you have a blank line at the end of your .call file.
>
> Nitesh Divecha wrote:
>   
>> Hello All,
>>
>> Something strange I found that my .call file is running twice...
>> Just after 60 sec it will run again, without any application invoking it.
>>
>> This is my .call file: -
>> =
>> Channel: SIP/xo-out/19097773456
>> Callerid: 9097773456
>> MaxRetries: 3
>> RetryTime: 30
>> WaitTime: 15
>> Context: custom-900
>> Extension: 900
>> Priority: 1
>>
>> I am running Asterisk 1.2.18 on CentOS 4.5.
>>
>> Anyone can help?
>>
>> Cheers,
>> Nitesh
>>
>>
>>
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>> 
>
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Re: [asterisk-users] .call file problem

[Nitesh Divecha]

Thanks Atis,

Yes and the .call executes fine... but after 60 seconds it executes 
again automatically without any application executing it.

Cheers,
Nitesh



Atis wrote:
> Is your .call file writable by asterisk?
>
> $ chmod 777 sample.call
>
> On 7/31/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:
>   
>> Hello All,
>>
>> Something strange I found that my .call file is running twice...
>> Just after 60 sec it will run again, without any application invoking it.
>>
>> This is my .call file: -
>> =
>> Channel: SIP/xo-out/19097773456
>> Callerid: 9097773456
>> MaxRetries: 3
>> RetryTime: 30
>> WaitTime: 15
>> Context: custom-900
>> Extension: 900
>> Priority: 1
>>
>> I am running Asterisk 1.2.18 on CentOS 4.5.
>>
>> Anyone can help?
>>
>> Cheers,
>> Nitesh
>>
>>
>>
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>> 
>
>
>   


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[asterisk-users] .call file problem

[Nitesh Divecha]

Hello All,

Something strange I found that my .call file is running twice...
Just after 60 sec it will run again, without any application invoking it.

This is my .call file: -
=
Channel: SIP/xo-out/19097773456
Callerid: 9097773456
MaxRetries: 3
RetryTime: 30
WaitTime: 15
Context: custom-900
Extension: 900
Priority: 1

I am running Asterisk 1.2.18 on CentOS 4.5.

Anyone can help?

Cheers,
Nitesh



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Re: [asterisk-users] PhpAgi call generation

[Nitesh Divecha]

Thanks Nasir,

That helped alot...

Cheers,
Nitesh


Nasir Iqbal wrote:
> Oh,
>
> you need Dial application instead of origination.
>
> so no need to AGI Script simply add
>
>
> the dialplan called for ".call" should look like this
>
> exten => yourexten,1,BackGround(your_menu_ivr)
> exten => yourexten,n,WaitExten()
>
> exten => 1,1,Dial(SIP/xo-out/$supervisor_num) ;for Supervisor
> exten => 2,1,Dial(SIP/xo-out/$manager_num) ;for Manager
> exten => 3,1,Voicemail(your_voice_mail_box)
>
>
> Regards
>
> Nasir Iqbal
>
>
> On Tue, 2007-07-31 at 12:21 -0400, Nitesh Divecha wrote:
>   
>> Thanks Nasir,
>>
>> By putting "'Exten'=> your_extensions_here" it will create a new channel 
>> to that extension, correct?
>>
>> What I want to do is to join two channels... Join the User A channel 
>> which is active with supervisor.
>>
>> Cheers,
>> Nitesh
>>
>>
>>
>> Nasir Iqbal wrote:
>> 
>>> Hi Nitesh,
>>>
>>> you are missing Extension
>>> try with
>>>
>>> $call = $asm->send_request('Originate',
>>> array('Channel'=>"SIP/xo-out/$supervisor_num",
>>> 'Context'=>'default',
>>> 'Exten'=> your_extensions_here,
>>> 'Priority'=>1,
>>> 'Callerid'=>$cid));
>>>
>>> or you must put an "s" extensions in your desired context in this case
>>> it is "default".
>>>
>>> Regards
>>>
>>> Nasir Iqbal
>>>
>>> On Tue, 2007-07-31 at 10:08 -0400, Nitesh Divecha wrote:
>>>   
>>>   
>>>> Hello All,
>>>>
>>>> Can anyone help me with this... This is what my program does: -
>>>>
>>>> 1) At certain time the system generates a ".call" and make a call to User 
>>>> A.
>>>>
>>>> 2) When User A picks up the phone call, system will play a menu select 
>>>> option.
>>>>a) Press 1 to call your supervisor.
>>>>b) Press 2 to call your manager.
>>>>c) Press 3 to leave a voice message.
>>>>
>>>> 3) When the User A press 1 to call his supervisor... The system has to 
>>>> put the User A on hold and place a call to the supervisor.
>>>>
>>>> 4) Once the supervisor picks up the call, User A has to be in session 
>>>> with his supervisor.
>>>>
>>>> Now I have already got part 1 and 2 done... but I am stuck with part 3 
>>>> and 4.
>>>>
>>>> This is how I generate my call to the supervisor: -
>>>> ===
>>>> if($asm->connect())
>>>> {
>>>> $call = $asm->send_request('Originate',
>>>> array('Channel'=>"SIP/xo-out/$supervisor_num",
>>>> 'Context'=>'default',
>>>> 'Priority'=>1,
>>>> 'Callerid'=>$cid));
>>>> $asm->disconnect();
>>>> }
>>>>
>>>> One the *CLI I do see the call, but its failing: -
>>>>
>>>> AGI Rx << STREAM FILE 
>>>> /var/spool/asterisk//tmp//text2wav_e08db16aede0af38ebb90a1c69ee19e3 "" 0
>>>> AGI Tx >> 200 result=0 endpos=26224
>>>>   == Parsing '/etc/asterisk/manager.conf': Found
>>>>   == Manager 'phpagi' logged on from 127.0.0.1
>>>>> Channel SIP/xo-out-08f8ae10 was answered.
>>>>   == Starting SIP/xo-out-08f8ae10 at default,,1 failed so falling back 
>>>> to exten 's'
>>>>   == Manager 'phpagi' logged off from 127.0.0.1
>>>> AGI Rx << STREAM FILE goodbye "" 0
>>>>
>>>> Can anyone put some light what I am missing here... Why the call is 
>>>> dropped on both end...?
>>>>
>>>> Cheers,
>>>> Nitesh
>>>>
>>>>
>>>>
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>>>> 
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>>>   
>
>
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Re: [asterisk-users] g729 setup help

[Nitesh Divecha]

Patrick,

Make sure you have install the g729 modules correctly as per the 
instructions and restarted Asterisk.

Other method is you can configure your Asterisk to do Pass-thru g729 
which you don't require to install any g729 license on Asterisk. As far 
as both of your phones has g729 installed Asterisk will just pass the 
traffic... Check out 
http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru

Cheers,
Nitesh



Patrick Fortin wrote:
> Hi
>
> I am trying to make this setup work
>
> phone1---g729---asterisk1---sip---asterisk2---g729---phone2
>
> I have tried several configurations but none worked
>
> I keep getting transcoding errors
>
> I have installed one g729 licence on each asterisk, but I can't verifiy 
> because the show g729 command is not available,
> I use 1.2.17
>
> Do I need 2 g729 licences per asterisk ?
>
> Do I need to register asterisk1 on asterisk2 and asterisk2 on asterisk1 ?
>
> Thanks
>
> Patrick
>
>
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Re: [asterisk-users] PhpAgi call generation

[Nitesh Divecha]

Thanks Nasir,

By putting "'Exten'=> your_extensions_here" it will create a new channel 
to that extension, correct?

What I want to do is to join two channels... Join the User A channel 
which is active with supervisor.

Cheers,
Nitesh



Nasir Iqbal wrote:
> Hi Nitesh,
>
> you are missing Extension
> try with
>
> $call = $asm->send_request('Originate',
> array('Channel'=>"SIP/xo-out/$supervisor_num",
> 'Context'=>'default',
> 'Exten'=> your_extensions_here,
> 'Priority'=>1,
> 'Callerid'=>$cid));
>
> or you must put an "s" extensions in your desired context in this case
> it is "default".
>
> Regards
>
> Nasir Iqbal
>
> On Tue, 2007-07-31 at 10:08 -0400, Nitesh Divecha wrote:
>   
>> Hello All,
>>
>> Can anyone help me with this... This is what my program does: -
>>
>> 1) At certain time the system generates a ".call" and make a call to User A.
>>
>> 2) When User A picks up the phone call, system will play a menu select 
>> option.
>>a) Press 1 to call your supervisor.
>>b) Press 2 to call your manager.
>>c) Press 3 to leave a voice message.
>>
>> 3) When the User A press 1 to call his supervisor... The system has to 
>> put the User A on hold and place a call to the supervisor.
>>
>> 4) Once the supervisor picks up the call, User A has to be in session 
>> with his supervisor.
>>
>> Now I have already got part 1 and 2 done... but I am stuck with part 3 
>> and 4.
>>
>> This is how I generate my call to the supervisor: -
>> ===
>> if($asm->connect())
>> {
>> $call = $asm->send_request('Originate',
>> array('Channel'=>"SIP/xo-out/$supervisor_num",
>> 'Context'=>'default',
>> 'Priority'=>1,
>> 'Callerid'=>$cid));
>> $asm->disconnect();
>> }
>>
>> One the *CLI I do see the call, but its failing: -
>>
>> AGI Rx << STREAM FILE 
>> /var/spool/asterisk//tmp//text2wav_e08db16aede0af38ebb90a1c69ee19e3 "" 0
>> AGI Tx >> 200 result=0 endpos=26224
>>   == Parsing '/etc/asterisk/manager.conf': Found
>>   == Manager 'phpagi' logged on from 127.0.0.1
>>> Channel SIP/xo-out-08f8ae10 was answered.
>>   == Starting SIP/xo-out-08f8ae10 at default,,1 failed so falling back 
>> to exten 's'
>>   == Manager 'phpagi' logged off from 127.0.0.1
>> AGI Rx << STREAM FILE goodbye "" 0
>>
>> Can anyone put some light what I am missing here... Why the call is 
>> dropped on both end...?
>>
>> Cheers,
>> Nitesh
>>
>>
>>
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[asterisk-users] PhpAgi call generation

[Nitesh Divecha]

Hello All,

Can anyone help me with this... This is what my program does: -

1) At certain time the system generates a ".call" and make a call to User A.

2) When User A picks up the phone call, system will play a menu select 
option.
   a) Press 1 to call your supervisor.
   b) Press 2 to call your manager.
   c) Press 3 to leave a voice message.

3) When the User A press 1 to call his supervisor... The system has to 
put the User A on hold and place a call to the supervisor.

4) Once the supervisor picks up the call, User A has to be in session 
with his supervisor.

Now I have already got part 1 and 2 done... but I am stuck with part 3 
and 4.

This is how I generate my call to the supervisor: -
===
if($asm->connect())
{
$call = $asm->send_request('Originate',
array('Channel'=>"SIP/xo-out/$supervisor_num",
'Context'=>'default',
'Priority'=>1,
'Callerid'=>$cid));
$asm->disconnect();
}

One the *CLI I do see the call, but its failing: -

AGI Rx << STREAM FILE 
/var/spool/asterisk//tmp//text2wav_e08db16aede0af38ebb90a1c69ee19e3 "" 0
AGI Tx >> 200 result=0 endpos=26224
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'phpagi' logged on from 127.0.0.1
   > Channel SIP/xo-out-08f8ae10 was answered.
  == Starting SIP/xo-out-08f8ae10 at default,,1 failed so falling back 
to exten 's'
  == Manager 'phpagi' logged off from 127.0.0.1
AGI Rx << STREAM FILE goodbye "" 0

Can anyone put some light what I am missing here... Why the call is 
dropped on both end...?

Cheers,
Nitesh



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[asterisk-users] AGI Que "Say Time"

[Nitesh Divecha]

Hello All,

I am almost done with my notifications system, but I am stuck with 
prompting the correct time.

I went over the "phpagi" doc's, on how to say a given time using "SAY 
TIME  .

According to http://www.voip-info.org/wiki/view/say+time it say  
is number of seconds elapsed since 00:00:00 on January 1, 1970, 
Coordinated Universal Time (UTC).

Do I have to compute my time based on 00:00:00 on January 1, 1970 and 
then it will prompt correct time?

What I am looking for is that, say the time on any given number of seconds.

Anyone can help?

Cheers,
Nitesh



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Re: [asterisk-users] Upgrade Procedure

[Nitesh Divecha]

Thanks Jared,

Does the same procedure works for updating Zaptel, Libpri, and Asterisks-Addons?

Cheers,
Nitesh




Jared Smith wrote:
> On Thu, 2007-07-19 at 10:53 -0400, Nitesh Divecha wrote:
>   
>> But is it possible to upgrade from Asterisk 1.2 to Asterisk 1.4?
>> I went over the UPGRADE.txt but it didn't explain much about 
>> uninstalling the old version and then install a new version.
>> 
>
> For the most part, you should simply be able to install Asterisk 1.4 on
> top of Asterisk 1.2.  The one place this won't work to well is with the
> Asterisk modules (usually located in /usr/lib/asterisk/modules).  I
> typically move those modules to a new location, then install Asterisk
> 1.4 over the top of Asterisk 1.2, and change my configuration files to
> match the new Asterisk 1.4 settings.
>
> Another common problem is that a couple of new items have been added to
> asterisk.conf, so I typically renamed asterisk.conf before installing
> 1.4, so that I get the new version of asterisk.conf as well.
>
>   


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Re: [asterisk-users] Upgrade Procedure

[Nitesh Divecha]

Thanks Yusuf,

But is it possible to upgrade from Asterisk 1.2 to Asterisk 1.4?
I went over the UPGRADE.txt but it didn't explain much about 
uninstalling the old version and then install a new version.

Cheers,
Nitesh


Yusuf wrote:
> X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more 
> information
> X-ECN Telecoms-MailScanner: Found to be clean
> X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED]
> X-Spam-Status: No
>
> Nitesh Divecha wrote:
>   
>> Hello All,
>>
>> I would like to upgrade my recently installed Asterisk 1.2.21.1 to 
>> Asterisk 1.4.8?
>>
>> My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 
>> 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux
>>
>> Is there any detail step by step procedure to uninstall the current 
>> version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 
>> 1.4.2?
>>
>> Cheers,
>> Nitesh
>> 
>
>
> Hi,
>
> there is an UPGRADE.txt file in each folder of asterisk, zaptel, etc.
> You now need to './configure' before 'make'.  Also check out 'make 
> menuselect' to select 
> which modules you need or don't.  Please check out the default configs first, 
> look in 
> asterisk-1.4.8/configs/
>
>
>
>   


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[asterisk-users] Upgrade Procedure

[Nitesh Divecha]

Hello All,

I would like to upgrade my recently installed Asterisk 1.2.21.1 to 
Asterisk 1.4.8?

My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 
05:25:07 EDT 2007 i686 i686 i386 GNU/Linux

Is there any detail step by step procedure to uninstall the current 
version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 
1.4.2?

Cheers,
Nitesh


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Re: [asterisk-users] Micros-Fidelio - billing in hotel

[Nitesh Divecha]

I have been looking for this solution for ages now... We have 5 hotels 
using Micros-Fidelio POS and I really like to integrate Asterisk PBX. 
Anything you can help...

Cheers,
Nitesh



Lee Jenkins wrote:
> Tomislav Parcina wrote:
>   
>> There is hotel application weary popular in Croatia - Micros-Fidelio. 
>> Now I need to connect Asterisk with this application for purpose of 
>> billing. Thing is that hotel would like to give customer one bill for 
>> every service that he used while he was in hotel.
>>
>> Has anybody connected Asterisk with Micros-Fidelio? As I understand this 
>> isn't some local developed application, it's something that is used 
>> world wide.
>>
>> Any informations are welcome.
>>
>>
>> 
>
> I wrote a middleware bridge (TCP => Serial) for Micros a 2 or 3 years 
> back and it was relatively simple.  This was the serial interface for 
> the 8700 standard.  If I remember correctly, it was a simple string that 
> was broken up into fixed length fields like char 1 through 10 was a 
> field and chars 11 through 15 was a field, etc.
>
> If you need help, email me off list and I'll look for that source code. 
>   Lucky for you it was written in pascal so its easy to read ;)
>
> Warm Regards,
>
> Lee
>
>
>
>
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Re: [asterisk-users] Error While Calling AGI

[Nitesh Divecha]

Thanks Russell...

I will apply it to my code and test it again... Will keep you updated...

Cheers,
Nitesh


Russell Bryant wrote:
> Russell Bryant wrote:
>   
>> Nitesh Divecha wrote:
>> 
>>> Found some strange problem while Asterisk trying to call the AGI file.
>>> If I pick up the call on the first attempt, it will execute my AGI file 
>>> properly.
>>> But if I don't pick up the call and let Asterisk call me again, it adds 
>>> StartRetry next to my AGI file name.
>>> Which will cause the AGI to fail to execute.
>>>   
>>  From a quick look at the code, it looks like you can resolve this problem 
>> by 
>> making sure that there is a newline at the end of your call file.  If there 
>> isn't, this problem will occur.
>>
>> I do consider it a bug that this can happen.  I'm going to go ahead and fix 
>> it. 
>>   But, that is what you can do to make it work until the fix makes it into a 
>> release.
>> 
>
> Here is the fix for this problem in case you would like to apply it to code 
> you 
> already have on your machine.
>
> http://lists.digium.com/pipermail/asterisk-commits/2007-July/014224.html
>
>   


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[asterisk-users] Error While Calling AGI

[Nitesh Divecha]

Hello All,
Please anyone can help me with this error...

Found some strange problem while Asterisk trying to call the AGI file.
If I pick up the call on the first attempt, it will execute my AGI file 
properly.
But if I don't pick up the call and let Asterisk call me again, it adds 
"StartRetry: 3700 1 (1182971439)" next to my AGI file name, which will 
cause the AGI to fail to execute.

-- Attempting call on SIP/5181 for application AGI(recordvoice.php) 
(Retry 1)
   -- Attempting call on SIP/5181 for application 
AGI(recordvoice.phpStartRetry: 3700 1 (1182971439)) (Retry 2)
   -- Attempting call on SIP/5181 for application 
AGI(recordvoice.phpStartRetry: 3700 1 (1182971439)) (Retry 3)
  > Channel SIP/08f39360 was answered.
  > Launching AGI(recordvoice.phpStartRetry: 3700 1 (1182971439)) on 
SIP/08f39360
   -- Launched AGI Script 
/var/lib/asterisk/agi-bin/recordvoice.phpStartRetry: 3700 1 (1182971439)
   -- AGI Script recordvoice.phpStartRetry: 3700 1 (1182971439) 
completed, returning 0

Can anyone help? By the way I am executing using *.call file.

File make.call: -
Channel: SIP/5181
MaxRetries: 3
RetryTime: 30
WaitTime: 15
Application: AGI
Data: recordvoice.php

Cheers,
Nitesh




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Re: [asterisk-users] .call file

[Nitesh Divecha]

Thanks Jerry,

But how can I access the Set variable in my AGI file?

Like I do for callerId $cidnum = $agi->request['agi_callerid'];

Is there any for Set?

Cheers,
Nitesh



Jerry Geis wrote:
> You can certainly use variables in the call file that get passed to the AGI.
> SetVar: MyVar=44
>
>
> jerry
>
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[asterisk-users] Error While Calling AGI

[Nitesh Divecha]

Hello All,

Found some strange problem while Asterisk trying to call the AGI file.
If I pick up the call on the first attempt, it will execute my AGI file 
properly.
But if I don't pick up the call and let Asterisk call me again, it adds 
StartRetry next to my AGI file name.
Which will cause the AGI to fail to execute.

-- Attempting call on SIP/5181 for application AGI(recordvoice.php) 
(Retry 1)
-- Attempting call on SIP/5181 for application 
AGI(*recordvoice.phpStartRetry: 3700 1 (1182971439)*) (Retry 2)
-- Attempting call on SIP/5181 for application 
AGI(*recordvoice.phpStartRetry: 3700 1 (1182971439)*) (Retry 3)
   > Channel SIP/08f39360 was answered.
   > Launching AGI(*recordvoice.phpStartRetry*: 3700 1 (1182971439)) 
on SIP/08f39360
-- Launched AGI Script 
/var/lib/asterisk/agi-bin/*recordvoice.phpStartRetry: 3700 1 (1182971439)*
-- AGI Script *recordvoice.phpStartRetry: 3700 1 (1182971439)* 
completed, returning 0

Can anyone help? By the way I am executing using *.call file.

File make.call: -
Channel: SIP/5181
MaxRetries: 3
RetryTime: 30
WaitTime: 15
Application: AGI
Data: recordvoice.php

Cheers,
Nitesh



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[asterisk-users] .call file

[Nitesh Divecha]

Hello All,

Is there any way to pass additional parameters while calling AGI from 
*.call file?

Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php

Something like Data: recordvoice.php?id=3453&name=asterisk

Cheers,
Nitesh



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Re: [asterisk-users] Nokia N95 + Dial Plan

[Nitesh Divecha]

Thanks Benny...
Let me give it a try...

Cheers,
Nitesh


Benny Amorsen wrote:
>>>>>> "ND" == Nitesh Divecha <[EMAIL PROTECTED]> writes:
>>>>>> 
>
> ND> Hello All, Recently I added some Nokia N95 customers and it worked
> ND> pretty good. Now the customers are complaining about the dialing
> ND> rules... They are used to dialing +12486543210 and +4479XX for
> ND> long distance calls.
>
> ND> Is there anyway to create a "+" sign dial plan which will allow
> ND> them to dial a number with "+" sign.
>
> You have your standard dialplan, usually something like:
>
>  exten => _X.,1,...
>
> You just put this in:
>
>  exten => _+.,1,Goto(00${EXTEN:1},1)
>
> And poof, all numbers starting with + get it replaced with 00.
>
> You may need to do more munching of course, and 00 may not be right
> for you.
>
>
> /Benny
>
>
>
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[asterisk-users] Nokia N95 + Dial Plan

[Nitesh Divecha]

Hello All,

Recently I added some Nokia N95 customers and it worked pretty good.
Now the customers are complaining about the dialing rules...
They are used to dialing +12486543210 and +4479XX for long distance 
calls.

Is there anyway to create a "+" sign dial plan which will allow them to 
dial a number with "+" sign.

Cheers,
Nitesh


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Re: [asterisk-users] PhpAgi call generation

[Nitesh Divecha]

Thanks Lee,

That really helped me to get my project started... I am in process of 
developing IVR based Notification System which is going to integrate 
with my IVR based Time clock system.

Notifications will be based on, if an employee is late to clock in, 
event should trigger and generate a .call file and call the supervisor 
and let him know that XYZ employee is late, do you want to inform an 
employee... etc...

Cheers,
Nitesh




Lee Jenkins wrote:
> Nitesh Divecha wrote:
>   
>> Is there any info on how to create .call files with some examples? And 
>> where to place this file? And how to initiate it..?
>>
>> Thanks man...
>>
>> Cheers,
>> Nitesh
>>
>>
>>
>> Christopher Robinson wrote:
>> 
>>> That should be pretty easy to do with a .call file.  The context that 
>>> you drop your called party off to will play the sounds and do the 
>>> transfer.  So really you need to concentrate on creating that context, 
>>> the .call files are very easy to generate.
>>>
>>>
>>> Nitesh Divecha wrote:
>>>   
>>>> Finally, this is what I was looking for... to generate a call.
>>>>
>>>> I have been working on my Time Clock application, where an employee will 
>>>> call into the system using his cellphone to clock in and clock out his 
>>>> hours. And it works perfect...
>>>>
>>>> Now I was looking for an option where or if an employee is late to clock 
>>>> in, the system has to generate a call and call the supervisor and inform 
>>>> him that XYZ employee is late and give an option to supervisor "Would 
>>>> you like to call XYZ employee, Press 1" and the system will call the XYZ 
>>>> employee and connect with the supervisor...
>>>>
>>>> Is it something feasible to do using the .call files? Or I am way too 
>>>> off...
>>>>
>>>> Cheers,
>>>> Nitesh
>>>>
>>>>
>>>> Christopher Robinson wrote:
>>>>   
>>>> 
>>>>> I've done this many times, also used the .call files.  If you don't need 
>>>>> your application to initiate the call the .call files are the better way 
>>>>> to go, otherwise it's a bit too much file management overhead.
>>>>>
>>>>> Here's some working code on our end.  In this case the Channel is 
>>>>> actually a context which makes the actual call, but I've used it both 
>>>>> ways.
>>>>>
>>>>> >>>>   require('PHPAGI/phpagi-asmanager.php');
>>>>>
>>>>>   $callid = 'Somebody';
>>>>>
>>>>>   $asm = new AGI_AsteriskManager();
>>>>>   if($asm->connect())
>>>>>   {
>>>>> $call = $asm->send_request('Originate',
>>>>> array('Channel'=>"LOCAL/[EMAIL PROTECTED]",
>>>>>   'Context'=>'called_party_context',
>>>>>   'Exten'=>'899',
>>>>>   'Timeout' => '1000',
>>>>>   'Async'=>'1',
>>>>>   'MaxRetries' => '5',
>>>>>   'RetryTime' => '5',
>>>>>   'Priority'=>1,
>>>>>   'Callerid'=>$callid));
>>>>> $asm->disconnect();
>>>>>   }
>>>>> ?>
>>>>>
>>>>>
>>>>> nik600 wrote:
>>>>>   
>>>>> 
>>>>>   
>>>>>> hi
>>>>>>
>>>>>> i'd like to write a simply application in php with phpAgi that:
>>>>>>
>>>>>> - connect to Asterisk
>>>>>> - call an external number using a Zap channel
>>>>>> - play a message
>>>>>>
>>>>>> here is some code:
>>>>>>
>>>>>> >>>>>
>>>>>> $asm = new AGI_AsteriskManager();
>>>>>>
>>>>>> if ($asm->connect()) {
>>>>>>
>>>>>> $asm->Originate("Zap/g1/1","number","default","1");
>>>>>>
>>>>>> /*
>>>>>> play message...
>>>>>> */
>>>>>> } else {
>>>>>> die("error\n");
>>>>>> }
>>>>>>
>>>>>> ?>
>>>>>>
>>>>>> But it doesn't work.
>>>>>> Is it possible to create a program like this?
>>>>>> thanks
>>>>>> 
>
> Sorry, I can't help you with PHP.  All my stuff is in pascal.  But here 
> is a link to call origination info:
> http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
>
> I did something a bit like what you're doing, but it was a script to 
> call into the system and generate a "broadcast" type message to a 
> different party.  Again, a bit different, but the elements are all the 
> same; call control, origination, database access, etc. Its in pascal, 
> but the syntax is very easy to understand and may give you an idea of 
> how program flow might be.
>
> http://www.leebo.dreamhosters.com/apscripts/msgcast/
>
>
>   


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[asterisk-users] How to Create Custom Context

[Nitesh Divecha]

Hello All,

Is there any way to write a custom context, where first it checks 
internally to see if the SIP User exist with same DID number, if it does 
route the call internally like calling from one extension to another 
extension, else pass the call to A2Billing to do the billing and use the 
default outbound trunks to terminate the call.

Reason for this is because I have generated my SIP User with real US DID 
and I would like to keep the cost minimum by not sending the local calls 
out on Trunk and receive it back...

For example,

[custom-a2billing]
exten => _XX,1,Answer
exten => _XX,2,Wait,2
exten => _XX,3,_ _ _ _ _ _ _ ; What to fill here to check for 
local calls and it not found send to A2Billing.php
exten => _XX,4,DeadAGI(a2billing.php|1)
exten => _XX,5,Wait,2
exten => _XX,6,Hangup

Any suggestions...?

Cheers,
Nitesh



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[asterisk-users] Asterisk RealTime

[Nitesh Divecha]

Hello All,

I manage to configure Asterisk RealTime and now it loads the SIP 
users/peers from MySQL DB. The table I am using is of A2Billing DB 
"cc_sip_buddies".

Now the only problem I am facing is incoming calls are failing... The 
ATA which is assigned this DID number is behind NAT and according to 
Olle's explanations he said "*there's no support for NAT keep-alives 
(qualify=) or voicemail indications* for these peers." 
http://www.voip-info.org/wiki/view/Asterisk+RealTime

So does this mean that I can not have any ATA behind NAT with this kind 
of setup? Below is the error message.

NOTE: The same setup used to work when I was using flat file config.

Here is my "extconfig.conf": -
[settings]
sipusers => mysql,mya2billing,cc_sip_buddies
sippeers => mysql,mya2billing,cc_sip_buddies

Here is my "res_mysql.conf": -
[general]
dbhost = 127.0.0.1
dbname = mya2billing
dbuser = billinguser
dbpass = 000eFm500F9E36
dbport = 3306
dbsock = /tmp/mysql.sock

When I do "sip show peers" on the *CLI I can see my SIP user: -
hyperion*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status   
2486543210/248654321  69.148.36.78 D   N  38813OK (20 ms)
1 sip peers [1 online , 0 offline]

Error message while receiving the call: -
-- AGI Script Executing Application: (DIAL) Options: 
(SIP/2486543210|60|HL(360:61000:3))
-- Limit Data for this call:
-- - timelimit = 360
-- - play_warning  = 61000
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq  = 3
-- - start_sound   = UNDEF
-- - warning_sound = timeleft
-- - end_sound = UNDEF
Jun 20 09:49:58 NOTICE[24952]: app_dial.c:1069 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
  a2billing.php|1|did: file:Class.A2Billing.php - line:634 - [CARD 
STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE 
username='2486543210']
-- AGI Script a2billing.php completed, returning 0

Any advice...

Cheers,
Nitesh



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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks everyone for the input...

In real world we can not ask the customers to dial 9, if they want to 
call another SIP user... and trust me its confusing for a customer 
also... meaning when to dial 9 and when to not...

We have a custom proprietary system which does this part very well... 
Before it sends the call on a Trunk it will check the DID, if it exists 
within the local system. If it does then it will just use IP to IP call, 
else send the call to Trunk...

I think its possible to do this by creating some basic dial plans... 
Same like creating local extensions.

Cheers,
Nitesh




John Novack wrote:
> Given that Asterisk is modeled on, in the telephone industry, an 
> obsolete PBX design, without many of the modern day hybrid features, and 
> only recently has any effort been made to provide buttons and lights for 
> "lines" ( Is that yet working in 1.4??) one would have to do some very 
> careful number parsing to not use a trunk digit.
>
> If every phone in the system had buttons and lights representing 
> external connections and internal connections on other button(s) ( 
> intercom ) this wouldn't be an issue.
> Most "legacy" systems have been able to do this for the last 20 years or so.
>
> John Novack
>
>
> Nitesh Divecha wrote:
>   
>> Thanks man,
>>
>> Is there any other way without dialing 9... it will be kinda pain for a 
>> customer to dial 9 every time and plus they need to know also...
>>
>> Is there any intelligent way to identify? if its a local SIP then don't 
>> route to Trunk else route to Trunk.
>>
>> Cheers,
>> Nitesh
>>
>>
>> Guillermo Salas M. wrote:
>>   
>> 
>>> On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote:
>>>   
>>> 
>>>   
>>>> Thanks man...
>>>>
>>>> So far everything worked as expected...
>>>>
>>>> How can I make internal calls stay within the PBX. For example, when
>>>> one 
>>>> SIP-Friend tries to call another SIP-Friend without sending the call
>>>> out 
>>>> on Trunk and receive it back. Same like dialing from one extension 
>>>> number to another extension.
>>>>
>>>> My SIP-Friends are using US DID numbers and I would like to keep the 
>>>> local calls within the network.
>>>>
>>>> Right now when I try to call other SIP-Friend, I get a message saying 
>>>> "The number you have dialer is currently not available"... while the 
>>>> SIP-Friend is registered.
>>>>
>>>> 
>>>>   
>>>> 
>>> Try dialing the number 9 before the sip/iax2 friend number.
>>>
>>> Regards,
>>>
>>>
>>>   
>>> 
>>>   
>>>> Cheers,
>>>> Nitesh 
>>>> 
>>>>   
>>>> 
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Re: [asterisk-users] PhpAgi call generation

[Nitesh Divecha]

Is there any info on how to create .call files with some examples? And 
where to place this file? And how to initiate it..?

Thanks man...

Cheers,
Nitesh



Christopher Robinson wrote:
> That should be pretty easy to do with a .call file.  The context that 
> you drop your called party off to will play the sounds and do the 
> transfer.  So really you need to concentrate on creating that context, 
> the .call files are very easy to generate.
>
>
> Nitesh Divecha wrote:
>> Finally, this is what I was looking for... to generate a call.
>>
>> I have been working on my Time Clock application, where an employee will 
>> call into the system using his cellphone to clock in and clock out his 
>> hours. And it works perfect...
>>
>> Now I was looking for an option where or if an employee is late to clock 
>> in, the system has to generate a call and call the supervisor and inform 
>> him that XYZ employee is late and give an option to supervisor "Would 
>> you like to call XYZ employee, Press 1" and the system will call the XYZ 
>> employee and connect with the supervisor...
>>
>> Is it something feasible to do using the .call files? Or I am way too 
>> off...
>>
>> Cheers,
>> Nitesh
>>
>>
>> Christopher Robinson wrote:
>>   
>>> I've done this many times, also used the .call files.  If you don't need 
>>> your application to initiate the call the .call files are the better way 
>>> to go, otherwise it's a bit too much file management overhead.
>>>
>>> Here's some working code on our end.  In this case the Channel is 
>>> actually a context which makes the actual call, but I've used it both ways.
>>>
>>> >>   require('PHPAGI/phpagi-asmanager.php');
>>>
>>>   $callid = 'Somebody';
>>>
>>>   $asm = new AGI_AsteriskManager();
>>>   if($asm->connect())
>>>   {
>>> $call = $asm->send_request('Originate',
>>> array('Channel'=>"LOCAL/[EMAIL PROTECTED]",
>>>   'Context'=>'called_party_context',
>>>   'Exten'=>'899',
>>>   'Timeout' => '1000',
>>>   'Async'=>'1',
>>>   'MaxRetries' => '5',
>>>   'RetryTime' => '5',
>>>   'Priority'=>1,
>>>   'Callerid'=>$callid));
>>> $asm->disconnect();
>>>   }
>>> ?>
>>>
>>>
>>> nik600 wrote:
>>>   
>>> 
>>>> hi
>>>>
>>>> i'd like to write a simply application in php with phpAgi that:
>>>>
>>>> - connect to Asterisk
>>>> - call an external number using a Zap channel
>>>> - play a message
>>>>
>>>> here is some code:
>>>>
>>>> >>>
>>>> $asm = new AGI_AsteriskManager();
>>>>
>>>> if ($asm->connect()) {
>>>>
>>>> $asm->Originate("Zap/g1/1","number","default","1");
>>>>
>>>> /*
>>>> play message...
>>>> */
>>>> } else {
>>>> die("error\n");
>>>> }
>>>>
>>>> ?>
>>>>
>>>> But it doesn't work.
>>>> Is it possible to create a program like this?
>>>> thanks
>>>>
>>>>   
>>>> 
>>>>   
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>>>   
>>> 
>>
>>
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>>   
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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks man,

Is there any other way without dialing 9... it will be kinda pain for a 
customer to dial 9 every time and plus they need to know also...

Is there any intelligent way to identify? if its a local SIP then don't 
route to Trunk else route to Trunk.

Cheers,
Nitesh


Guillermo Salas M. wrote:
> On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote:
>   
>> Thanks man...
>>
>> So far everything worked as expected...
>>
>> How can I make internal calls stay within the PBX. For example, when
>> one 
>> SIP-Friend tries to call another SIP-Friend without sending the call
>> out 
>> on Trunk and receive it back. Same like dialing from one extension 
>> number to another extension.
>>
>> My SIP-Friends are using US DID numbers and I would like to keep the 
>> local calls within the network.
>>
>> Right now when I try to call other SIP-Friend, I get a message saying 
>> "The number you have dialer is currently not available"... while the 
>> SIP-Friend is registered.
>>
>> 
>
> Try dialing the number 9 before the sip/iax2 friend number.
>
> Regards,
>
>
>   
>> Cheers,
>> Nitesh 
>> 


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Re: [asterisk-users] PhpAgi call generation

[Nitesh Divecha]

Finally, this is what I was looking for... to generate a call.

I have been working on my Time Clock application, where an employee will 
call into the system using his cellphone to clock in and clock out his 
hours. And it works perfect...

Now I was looking for an option where or if an employee is late to clock 
in, the system has to generate a call and call the supervisor and inform 
him that XYZ employee is late and give an option to supervisor "Would 
you like to call XYZ employee, Press 1" and the system will call the XYZ 
employee and connect with the supervisor...

Is it something feasible to do using the .call files? Or I am way too 
off...

Cheers,
Nitesh


Christopher Robinson wrote:
> I've done this many times, also used the .call files.  If you don't need 
> your application to initiate the call the .call files are the better way 
> to go, otherwise it's a bit too much file management overhead.
>
> Here's some working code on our end.  In this case the Channel is 
> actually a context which makes the actual call, but I've used it both ways.
>
>require('PHPAGI/phpagi-asmanager.php');
>
>   $callid = 'Somebody';
>
>   $asm = new AGI_AsteriskManager();
>   if($asm->connect())
>   {
> $call = $asm->send_request('Originate',
> array('Channel'=>"LOCAL/[EMAIL PROTECTED]",
>   'Context'=>'called_party_context',
>   'Exten'=>'899',
>   'Timeout' => '1000',
>   'Async'=>'1',
>   'MaxRetries' => '5',
>   'RetryTime' => '5',
>   'Priority'=>1,
>   'Callerid'=>$callid));
> $asm->disconnect();
>   }
> ?>
>
>
> nik600 wrote:
>   
>> hi
>>
>> i'd like to write a simply application in php with phpAgi that:
>>
>> - connect to Asterisk
>> - call an external number using a Zap channel
>> - play a message
>>
>> here is some code:
>>
>> >
>> $asm = new AGI_AsteriskManager();
>>
>> if ($asm->connect()) {
>>
>> $asm->Originate("Zap/g1/1","number","default","1");
>>
>> /*
>> play message...
>> */
>> } else {
>> die("error\n");
>> }
>>
>> ?>
>>
>> But it doesn't work.
>> Is it possible to create a program like this?
>> thanks
>>
>>   
>> 
>
>
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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks man...

So far everything worked as expected...

How can I make internal calls stay within the PBX. For example, when one 
SIP-Friend tries to call another SIP-Friend without sending the call out 
on Trunk and receive it back. Same like dialing from one extension 
number to another extension.

My SIP-Friends are using US DID numbers and I would like to keep the 
local calls within the network.

Right now when I try to call other SIP-Friend, I get a message saying 
"The number you have dialer is currently not available"... while the 
SIP-Friend is registered.

Cheers,
Nitesh




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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Strange...
Got it working now... I can receive incoming call...

Changed following parameters in additional_a2billing_sip.conf of the DID 
to: -

qualify=yes
canreinvite=no

Cheers,
Nitesh



Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote:
>   
>> When I call from my cell to the above DID, it hits on the Asterisk and
>> I 
>> see A2Billing trying to call SIP/2486543210, but it fails because 
>> Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No 
>> route to destination) ". 
>> 
>
> I know it, but the error is saying that you don't have one 2486543210
> user registred.
>
> Show us the output of:
>
> sip show peers
>
> Regards,
>
>   


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Here is my "sip show peers"

hyperion*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status   
2486543210/2486543210  86.14.22.128 D   N  61547LAGGED 
(66 ms)

Now here is the catch, before it used to show the status OK but now its 
showing LAGGED.
Dunno what does that means... Any suggestions...

Cheers,
Nitesh




Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote:
>   
>> When I call from my cell to the above DID, it hits on the Asterisk and
>> I 
>> see A2Billing trying to call SIP/2486543210, but it fails because 
>> Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No 
>> route to destination) ". 
>> 
>
> I know it, but the error is saying that you don't have one 2486543210
> user registred.
>
> Show us the output of:
>
> sip show peers
>
> Regards,
>
>   


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

2486543210 is my SIP-Friend which I created manually and associated with 
one of the card number.
My ATA is registered to Asterisk using the about DID Number.
So I want when I call the above number, it should ring on the ATA.
When I call from my cell to the above DID, it hits on the Asterisk and I 
see A2Billing trying to call SIP/2486543210, but it fails because 
Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No 
route to destination) ".

Any suggestion...

Cheers,
Nitesh





Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 17:01 -0400, Nitesh Divecha wrote:
>   
>> Thanks man... That really helped me to move couple of steps. Now I see
>> the incoming calls are going in proper direction... I know I am still
>> missing a small piece here... I did ADD the Destination as a
>> SIP/2486543210, assigned the card number, enabled VOIP_CALL, and
>> enabled Active. 
>>
>> 
>
>
> 2486543210 is your card number?
>
>
>   
>> When I dial the DID number, on the *CLI it shows the following: -
>>
>> a2billing.php|1|did: bug
>> -- AGI Script Executing Application: (DIAL) Options:
>> (SIP/2486543210|60|HL(360:61000:3))
>> -- Limit Data for this call: 
>> -- - timelimit = 360
>> -- - play_warning  = 61000
>> -- - play_to_caller= yes
>> -- - play_to_callee= no
>> -- - warning_freq  = 3
>> -- - start_sound   = UNDEF
>> -- - warning_sound = timeleft 
>> -- - end_sound = UNDEF
>> Destroying call '[EMAIL PROTECTED]'
>> Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable
>> to create channel of type 'SIP' (cause 3 - No route to destination) 
>>   == Everyone is busy/congested at this time (1:0/0/1)
>>
>> 
>
> I think that 2486543210 is not a customer, card number or SIP/IAX2
> friend, maybe is PSTN number. To redirect the call to any PSTN number
> you must need to set "voip call" to inactive and set the destination
> number to 2486543210.
>
>
>   
>> I bet I am missing something in extension.conf correct? I dont see any
>> examples in my package.
>>
>> 
>
>
> The context is fine don't worry about it.
>
>
>   
>> Any suggestion... Thanks once again... 
>> 
>
>
> Regards,
>
>   


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks man... That really helped me to move couple of steps. Now I see the
incoming calls are going in proper direction... I know I am still missing a
small piece here... I did ADD the Destination as a SIP/2486543210, assigned
the card number, enabled VOIP_CALL, and enabled Active.

When I dial the DID number, on the *CLI it shows the following: -

a2billing.php|1|did: bug
   -- AGI Script Executing Application: (DIAL) Options:
(SIP/2486543210|60|HL(360:61000:3))
   -- Limit Data for this call:
   -- - timelimit = 360
   -- - play_warning  = 61000
   -- - play_to_caller= yes
   -- - play_to_callee= no
   -- - warning_freq  = 3
   -- - start_sound   = UNDEF
   -- - warning_sound = timeleft
   -- - end_sound = UNDEF
Destroying call '[EMAIL PROTECTED]'
Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
 == Everyone is busy/congested at this time (1:0/0/1)

I bet I am missing something in extension.conf correct? I dont see any
examples in my package.

Any suggestion... Thanks once again...

Cheers,
Nitesh






On 6/15/07, Guillermo Salas M. <[EMAIL PROTECTED]> wrote:


On Fri, 2007-06-15 at 14:20 -0400, Nitesh Divecha wrote:
> You said change the context for SIP Customers to
> "context=a2billing-did", do I have to create this context or
> A2Billing
> will generate by itself?
>


The a2billing package comes with some examples, you must have to create
the a2billing-did context :

[a2billing-did]
exten => _X.,1,NoOp,${CALLERID(all)}
exten => _X.,2,DeadAGI(a2billing.php|1|did)
exten => _X.,3,Hangup()

This will be the context for your DID provider and not for your
customers.

Check this link for more information:

http://forum.asterisk2billing.org/viewtopic.php?t=1784


Cheers!

--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks Man...

Do I need to change my context in sip.conf to "context=a2billing" or 
should I leave it to "context=default"?

You said change the context for SIP Customers to 
"context=a2billing-did", do I have to create this context or A2Billing 
will generate by itself?

Cheers,
Nitesh



Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 12:19 -0400, Nitesh Divecha wrote:
>   
>> Thanks everyone,
>>
>> OK, I got everything working... I manage to create a SIP Customer with a 
>> real DID number and configured an ATA with the DID number. ATA can login 
>> and can make calls out without any issues.
>>
>> But incoming calls are failing... As soon as the call hits Asterisk, 
>> A2Billing script runs and ask for PIN Number... I checked the context 
>> for my DID it shows "context=a2billing" and under sip.conf 
>> "context=a2billing".
>>
>> If I change the default context under sip.conf to "context=default", 
>> then the calls are failing... meaning I do not get any response back, 
>> but on *CLI debug show that its failing to look for the DID number. 
>> Well, I know this is due to my DID is in  "context=a2billing".
>>
>> Anyone can suggest how can I fix this... I want to ring my incoming to 
>> that ATA which has DID assigned.
>> 
>
> You need to setup the DID on the DID section of a2billing.
>
> First create one SIP/IAX2 configuration for your DID provider and assign
> the context a2billing-did.
>
> Later on the DID section, add the DID Provider, add the DID number and
> asign one destination to the DID (your ata card number) or any SIP
> extension enabling the "voip call" radius button.
>
> Try it.
>
> Regards,
>
>
>   


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks everyone,

OK, I got everything working... I manage to create a SIP Customer with a 
real DID number and configured an ATA with the DID number. ATA can login 
and can make calls out without any issues.

But incoming calls are failing... As soon as the call hits Asterisk, 
A2Billing script runs and ask for PIN Number... I checked the context 
for my DID it shows "context=a2billing" and under sip.conf 
"context=a2billing".

If I change the default context under sip.conf to "context=default", 
then the calls are failing... meaning I do not get any response back, 
but on *CLI debug show that its failing to look for the DID number. 
Well, I know this is due to my DID is in  "context=a2billing".

Anyone can suggest how can I fix this... I want to ring my incoming to 
that ATA which has DID assigned.

Cheers,
Nitesh







Guillermo Salas M. wrote:
> On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote:
>   
>> Hello All,
>>
>> I got one quick question on A2Billing.
>>
>> Specs: -
>> - A2Billing v1.3
>> - OS CentOS 4.5
>> - Asterisk 1.2
>> - Zaptel 1.2
>>
>> Did the installation and everything is working as it suppose to...
>>
>> Using the A2Billing documentation, I created the RateCard, SIP Trunks, 
>> and SIP Customers. I was also able to login using XLite Dialer and was 
>> able to call out to my SIP Trunk also.
>>
>> Now how can I remove the IVR Prompt... Meaning from my XLite dialer I 
>> want to dial directly and let A2Billing do the billing part. Right now 
>> is something like when I dial any number from XLite, A2Billing script is 
>> invoked and it will announce "You have XXX amount, please enter the 
>> number you wish to call followed by #". And then I have to enter the 
>> number again and then the call is initiated... Its kinda annoying to do 
>> that every time you want to call.
>>
>> Is there anyway to modify config some where, so it will do the billing 
>> in background when the phone call is hangup.
>>
>> 
>
>
> Yes, is possible using the a2billing.conf file in the right way.
>
> I don't have the v1.3 installed, but in the previous release 1.2.3 you
> must have to modify :
>
> use_dnid=YES
> number_try=1
> say_balance_after_auth=NO
> say_balance_after_call=NO
> say_rateinitial=NO
> say_timetocall=NO
>
> Regards,
>
>   


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

That was easy... Thanks a million man...
Dunno what I was thinking and went too far writing custom scripts...

Cheers,
Nitesh



Guillermo Salas M. wrote:

On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote:
  

Hello All,

I got one quick question on A2Billing.

Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2

Did the installation and everything is working as it suppose to...

Using the A2Billing documentation, I created the RateCard, SIP Trunks, 
and SIP Customers. I was also able to login using XLite Dialer and was 
able to call out to my SIP Trunk also.


Now how can I remove the IVR Prompt... Meaning from my XLite dialer I 
want to dial directly and let A2Billing do the billing part. Right now 
is something like when I dial any number from XLite, A2Billing script is 
invoked and it will announce "You have XXX amount, please enter the 
number you wish to call followed by #". And then I have to enter the 
number again and then the call is initiated... Its kinda annoying to do 
that every time you want to call.


Is there anyway to modify config some where, so it will do the billing 
in background when the phone call is hangup.






Yes, is possible using the a2billing.conf file in the right way.

I don't have the v1.3 installed, but in the previous release 1.2.3 you
must have to modify :

use_dnid=YES
number_try=1
say_balance_after_auth=NO
say_balance_after_call=NO
say_rateinitial=NO
say_timetocall=NO

Regards,

  


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[asterisk-users] Que on A2Billing

[Nitesh Divecha]

Hello All,

I got one quick question on A2Billing.

Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2

Did the installation and everything is working as it suppose to...

Using the A2Billing documentation, I created the RateCard, SIP Trunks, 
and SIP Customers. I was also able to login using XLite Dialer and was 
able to call out to my SIP Trunk also.


Now how can I remove the IVR Prompt... Meaning from my XLite dialer I 
want to dial directly and let A2Billing do the billing part. Right now 
is something like when I dial any number from XLite, A2Billing script is 
invoked and it will announce "You have XXX amount, please enter the 
number you wish to call followed by #". And then I have to enter the 
number again and then the call is initiated... Its kinda annoying to do 
that every time you want to call.


Is there anyway to modify config some where, so it will do the billing 
in background when the phone call is hangup.


Cheers,
Nitesh

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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

What does codec has to do...? I am using G729a

Cheers,
Nitesh



ram wrote:



On 5/26/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:


Thanks Shanon and everyones input...

Finally, got the application working as planned with PHPAGI...

Now the only draw back is the voice... I am using text2wav to
prompt all
the questions, but the voice is creepy...

Is their any easier way to replace the text2wav voice with proper
recorded female voice?

Please advice...

 
 
what codec are you using
 
ram
 



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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Thanks Shanon and everyones input...

Finally, got the application working as planned with PHPAGI...

Now the only draw back is the voice... I am using text2wav to prompt all 
the questions, but the voice is creepy...


Is their any easier way to replace the text2wav voice with proper 
recorded female voice?


Please advice...

Cheers,
Nitesh









Shanon Swafford wrote:

I was messing with something similar one day for a trucking company to track
progress of their drivers.

It is HIGHLY beta, but should get you started:


## extensions.conf ###
exten => s,1,NoOp(FXO Line is Ringing : ${CALLERID(all)})
exten => s,n,NoOp(${CALLERID(all)})
exten => s,n,NoOp(${CALLERID(num)})
exten => s,n,NoOp(${CALLERID(name)})
exten => s,n,GotoIf($["${CALLERID(num)}"="9728311600"]?agitest|s|1)
exten => s,n,GotoIf($["${CALLERID(num)}"="200"]?agitest|s|1)

[agitest]
exten => s,1,AGI(test.php)
exten => s,n,Answer
exten => s,n,Background(shanon-welcome) ; "Thanks for calling press
1 for sales, 2 for support, ..."
exten => s,n,WaitExten




###test.php###
answer();

  $cidnum = $agi->request['agi_callerid'];
  $cidname = $agi->request['agi_calleridname'];

  $agi->text2wav("Hello $cidname");
  $agi->text2wav('We are testing so please call our cell phones.  ');

  $test = 0;
  while ( $test <> 1 ) {
$agi->text2wav("Enter your Order Number");
$load_num = $agi->get_data('beep', 3000, 6);
$tmp = strsplit($load_num);
$mydata = "";
foreach ($tmp as $value) {
  $mydata .= $value . "";
}
$agi->text2wav("You entered $mydata.  Enter 1 if this is correct");
$test = $agi->get_data('beep', 3000, 1);

$agi->conlog("Customer Entered: $test");
 }

/* Add code here to insert $test into a database */

  $agi->text2wav('Goodbye');
//  $agi->hangup();



function strsplit($str, $l=1) {
   do {$ret[]=substr($str,0,$l); $str=substr($str,$l); }
   while($str != "");
   return $ret;
}
?>


Regards,
Shanon
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh Divecha
Sent: Thursday, May 24, 2007 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Time Card


Thanks for your reply,

The basic system would work as follows: -

Method 1
===
An employee would call in to the system and a welcome message is 
prompted. After that a employee is asked to enter the employee ID and 
PIN number and once verified Employee ID, Caller ID, and time of day is 
stored into MySQL DB. By end of the day employee will call in again to 
logout from the system and same information is stored into the DB.


Method 2
===
This time employee is verified with Caller ID, so the employee ID and 
PIN number is skipped and time of day is logged into the DB.


Is it possible?

Thanks,
Nitesh







ram wrote:
  
On 5/24/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:


Hello All,

I have been looking for this solution for quite sometimes
"Asterisk Time
Card System". I found some discussion from Digium forum but not quite
helpful.

 
 
Hi
 
what is the mean of time card system ?
 
is this kind of attendent system ?
 
kindly give some more details
 
ram



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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Thanks alot Shanon... That helped me to kick start my work...

Cheers,
Nitesh




Shanon Swafford wrote:

I was messing with something similar one day for a trucking company to track
progress of their drivers.

It is HIGHLY beta, but should get you started:


## extensions.conf ###
exten => s,1,NoOp(FXO Line is Ringing : ${CALLERID(all)})
exten => s,n,NoOp(${CALLERID(all)})
exten => s,n,NoOp(${CALLERID(num)})
exten => s,n,NoOp(${CALLERID(name)})
exten => s,n,GotoIf($["${CALLERID(num)}"="9728311600"]?agitest|s|1)
exten => s,n,GotoIf($["${CALLERID(num)}"="200"]?agitest|s|1)

[agitest]
exten => s,1,AGI(test.php)
exten => s,n,Answer
exten => s,n,Background(shanon-welcome) ; "Thanks for calling press
1 for sales, 2 for support, ..."
exten => s,n,WaitExten




###test.php###
answer();

  $cidnum = $agi->request['agi_callerid'];
  $cidname = $agi->request['agi_calleridname'];

  $agi->text2wav("Hello $cidname");
  $agi->text2wav('We are testing so please call our cell phones.  ');

  $test = 0;
  while ( $test <> 1 ) {
$agi->text2wav("Enter your Order Number");
$load_num = $agi->get_data('beep', 3000, 6);
$tmp = strsplit($load_num);
$mydata = "";
foreach ($tmp as $value) {
  $mydata .= $value . "";
}
$agi->text2wav("You entered $mydata.  Enter 1 if this is correct");
$test = $agi->get_data('beep', 3000, 1);

$agi->conlog("Customer Entered: $test");
 }

/* Add code here to insert $test into a database */

  $agi->text2wav('Goodbye');
//  $agi->hangup();



function strsplit($str, $l=1) {
   do {$ret[]=substr($str,0,$l); $str=substr($str,$l); }
   while($str != "");
   return $ret;
}
?>


Regards,
Shanon
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh Divecha
Sent: Thursday, May 24, 2007 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Time Card


Thanks for your reply,

The basic system would work as follows: -

Method 1
===
An employee would call in to the system and a welcome message is 
prompted. After that a employee is asked to enter the employee ID and 
PIN number and once verified Employee ID, Caller ID, and time of day is 
stored into MySQL DB. By end of the day employee will call in again to 
logout from the system and same information is stored into the DB.


Method 2
===
This time employee is verified with Caller ID, so the employee ID and 
PIN number is skipped and time of day is logged into the DB.


Is it possible?

Thanks,
Nitesh







ram wrote:
  
On 5/24/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:


Hello All,

I have been looking for this solution for quite sometimes
"Asterisk Time
Card System". I found some discussion from Digium forum but not quite
helpful.

 
 
Hi
 
what is the mean of time card system ?
 
is this kind of attendent system ?
 
kindly give some more details
 
ram



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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Thanks Bruce,

If possible could you share your code...? I just need an idea how to 
integrate and store info in DB.


Cheers,
Nitesh



Bruce Reeves wrote:
This can be accomplished by writing an IVR to prompt and then using 
AGI or dialplan commands the query strings can be executed. I have a 
setup like this for a inegrating a in house time keeping system with 
asterisk.


On 5/24/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:


Thanks for your reply,

The basic system would work as follows: -

Method 1
===
An employee would call in to the system and a welcome message is
prompted. After that a employee is asked to enter the employee ID and
PIN number and once verified Employee ID, Caller ID, and time of
day is
stored into MySQL DB. By end of the day employee will call in again to
logout from the system and same information is stored into the DB.

Method 2
===
This time employee is verified with Caller ID, so the employee ID and
PIN number is skipped and time of day is logged into the DB.

Is it possible?

Thanks,
Nitesh







ram wrote:
>
>
    > On 5/24/07, *Nitesh Divecha* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>
> <mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>>> wrote:
>
> Hello All,
>
> I have been looking for this solution for quite sometimes
> "Asterisk Time
> Card System". I found some discussion from Digium forum but
not quite
> helpful.
>
>
>
> Hi
>
> what is the mean of time card system ?
>
> is this kind of attendent system ?
>
> kindly give some more details
>
> ram
>

>
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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--
Bruce Reeves
Nortex Networks


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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Thanks David,

Is it possible if you could share your code...
All I need is just an idea and develop my own.

Cheers,
Nitesh



David Gomillion wrote:
On 5/24/07, *Alex Balashov* <[EMAIL PROTECTED] 
> wrote:



This is all definitely possible by using Asterisk database
interfaces, but
I cannot find an existing implementation of something of this nature.

It is an unusual and clever application of Asterisk.  :-)


Don't know how unusual. When I do contract work, most of the jobs I do 
have a phone number to log in and out thru.


By the way, when I wrote the module, I cheated and used a System call 
(although I would use the TrySystem if I were to do it again) and 
called a very simple PHP script. Oh, and I authenticated within the 
dialplan so that I could easily play useful error messages without 
checking the returned value of the PHP script.


Not the best system, but it worked in my testing.




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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Thanks Mike,

Will look into Asterisk AGI...

Cheers,
Nitesh



Mike Clark wrote:

Nitesh Divecha wrote:
  

Thanks for your reply,

The basic system would work as follows: -

Method 1
===
An employee would call in to the system and a welcome message is
prompted. After that a employee is asked to enter the employee ID and
PIN number and once verified Employee ID, Caller ID, and time of day is
stored into MySQL DB. By end of the day employee will call in again to
logout from the system and same information is stored into the DB.

Method 2
===
This time employee is verified with Caller ID, so the employee ID and
PIN number is skipped and time of day is logged into the DB.

Is it possible?

Thanks,
Nitesh








Nitesh:

This would be pretty easy using AGI. We haven't done time and
attendance, but have implemented some reasonably complex IVR payment
systems integrating with MySQL. Many others have done similar and even
more extensive applications in this manner.

Google "Asterisk AGI" and this should get you started.

Mike Clark
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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

David,

You are correct... thats the whole scenario to simplify running payroll...

I am planning to do three level of verifications which will make sure 
the employee is in right location, so he is not spoofing anything...


1) Verify by Employee ID and PIN.
2) Verify by Location ID. This will be printed at location site.
3) Verify by token ID, generated by http://www.mypw.com/
4) Login the time or Logout the employee.

If the Caller is calling from the registered Caller ID, then step 1 will 
be ignored. Kinda like Caller ID authentication.


Thanks,
Nitesh








David Gomillion wrote:



On 5/24/07, *Alex Balashov* <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:


On Thu, 24 May 2007, Nitesh Divecha wrote:

> I have been looking for this solution for quite sometimes
"Asterisk Time
> Card System". I found some discussion from Digium forum but not
quite
> helpful.

   Are you by chance referring to chipsets that provide hardware
timing /
Real-Time Clock functionality used by Asterisk?


Unless I'm very much mistaken, he's referring to a Time and Attendance 
system. The idea is to capture times that a person clocks in and when 
the person clocks out, to simplify running payroll.
 





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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Thanks David,

Any code you can share... I just need a kick start...

Nitesh



David Gomillion wrote:
On 5/24/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:


Thanks for your reply,

The basic system would work as follows: -

Method 1
===
An employee would call in to the system and a welcome message is
prompted. After that a employee is asked to enter the employee ID and
PIN number and once verified Employee ID, Caller ID, and time of
day is
stored into MySQL DB. By end of the day employee will call in again to
logout from the system and same information is stored into the DB.

Method 2
===
This time employee is verified with Caller ID, so the employee ID and
PIN number is skipped and time of day is logged into the DB.

Is it possible?

Thanks,
Nitesh


Anything is possible. But I haven't seen one off-the-shelf. It really 
won't be a big deal to write, though. We created a timeclock 
application and toyed with allowing people to clock in via phone, and 
I even wrote the extension logic, but we opted to not enable it 
because we don't trust our employees that much.


This was years ago, when we were running pre-1.0 code. We've switched 
servers a few times, so the logic is long gone, but it only took an 
afternoon to write and debug.
 




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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Alex,

No, I don't refer to hardware timing... It is just a Unix time stamp as 
used by CDR's.


Thanks,
Nitesh



Alex Balashov wrote:

On Thu, 24 May 2007, Nitesh Divecha wrote:

I have been looking for this solution for quite sometimes "Asterisk 
Time Card System". I found some discussion from Digium forum but not 
quite helpful.


  Are you by chance referring to chipsets that provide hardware timing 
/ Real-Time Clock functionality used by Asterisk?


--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Thanks for your reply,

The basic system would work as follows: -

Method 1
===
An employee would call in to the system and a welcome message is 
prompted. After that a employee is asked to enter the employee ID and 
PIN number and once verified Employee ID, Caller ID, and time of day is 
stored into MySQL DB. By end of the day employee will call in again to 
logout from the system and same information is stored into the DB.


Method 2
===
This time employee is verified with Caller ID, so the employee ID and 
PIN number is skipped and time of day is logged into the DB.


Is it possible?

Thanks,
Nitesh







ram wrote:



On 5/24/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:


Hello All,

I have been looking for this solution for quite sometimes
"Asterisk Time
Card System". I found some discussion from Digium forum but not quite
helpful.

 
 
Hi
 
what is the mean of time card system ?
 
is this kind of attendent system ?
 
kindly give some more details
 
ram



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[asterisk-users] Asterisk Time Card

[Nitesh Divecha]

Hello All,

I have been looking for this solution for quite sometimes "Asterisk Time 
Card System". I found some discussion from Digium forum but not quite 
helpful.


Can anyone redirect me to the correct path?

Thanks,
Nitesh

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[asterisk-users] Asterisk Clusters

[Nitesh Divecha]

Hello All,

I need to implement a clustered PBX System where parent * is connected 
to one of the outbound carrier and other child * will register to parent 
*. Reason for this implementation is because some of the child * are 
behind NAT. Parent * is on Public IP Address and its connected to 
outbound carrier. Child * will only send out long distances calls to 
Parent * to terminate, rest are internal calls.


Now which is the best way to implement this type of scenario... DUNDi? 
or custom context?


Thanks,
Nitesh


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[asterisk-users] Asterisk + Hotel Management System

[Nitesh Divecha]

Hello All,

By any chance anyone has setup Asterisk PBX for big Hotels using Hotel 
Management System from "Micros System Inc." URL: 
http://www.micros.com/Industries/HotelsAndResorts/


Or any other solutions are welcome?

Thanks,
Nitesh

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[asterisk-users] Asterisk behind NAT

[Nitesh Divecha]

Hello All,

Has anyone implemented Asterisk behind D-Link Router?
Got one pain in butt customer who wants to setup * system behind D-Link 
router model DI-624?


Can anyone share their conf?

Thanks,
Nitesh


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Re: [asterisk-users] CAS signalling conflicts with Clear channel

[Nitesh Divecha]

Vieri,

Make sure you are loading the digital card first and then analog card.
I had the same problem and Digium engineers helped me out.

Cheers,
Nitesh


Vieri wrote:

--- David Gomillion <[EMAIL PROTECTED]> wrote:

  

On 5/21/07, Vieri <[EMAIL PROTECTED]> wrote:


Hi,

My asterisk server was working with a 4-FXO analog
card (TDM400P).

I recently added two digital cards: a TE120P (1
  

PRI)


and a B410P (4 BRI).

The B410P is still unconfigured but inserted in a
  

PCI


slot.

The TE120P's jumper is set to E1 as it will
  

connect to


a commercial PBX's PRI card also configured as E1.

My analog channels used to be 1-4 but since I
  

added


the new cards I changed them to 101-104.
  

I could be wrong here, but I don't think you get to
arbitrarily make up what
the channel numbers. At least I've never done that;
I let the first channel
be 1, second one 2, etc, through all of the cards,
based on loading order of
the PCI cards. And are you sure about the loading
order of the cards?



I'm sure you're right because the following yields no
error:

# misdn-init stop
# rmmod wctdm
# rmmod xpp
# rmmod wcte12xp
# rmmod zaptel
# modprobe -a zaptel
# modprobe -a wcte12xp
# ztcfg -v

Zaptel Configuration
==

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet
(DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)

31 channels configured.

Changing signalling on channel 1 from Unused to Clear
channel
Changing signalling on channel 2 from Unused to Clear
channel
Changing signalling on channel 3 from Unused to Clear
channel
Changing signalling on channel 4 from Unused to Clear
channel
Changing signalling on channel 5 from Unused to Clear
channel
Changing signalling on channel 6 from Unused to Clear
channel
Changing signalling on channel 7 from Unused to Clear
channel
Changing signalling on channel 8 from Unused to Clear
channel
Changing signalling on channel 9 from Unused to Clear
channel
Changing signalling on channel 10 from Unused to Clear
channel
Changing signalling on channel 11 from Unused to Clear
channel
Changing signalling on channel 12 from Unused to Clear
channel
Changing signalling on channel 13 from Unused to Clear
channel
Changing signalling on channel 14 from Unused to Clear
channel
Changing signalling on channel 15 from Unused to Clear
channel
Changing signalling on channel 16 from Unused to HDLC
with FCS check
Changing signalling on channel 17 from Unused to Clear
channel
Changing signalling on channel 18 from Unused to Clear
channel
Changing signalling on channel 19 from Unused to Clear
channel
Changing signalling on channel 20 from Unused to Clear
channel
Changing signalling on channel 21 from Unused to Clear
channel
Changing signalling on channel 22 from Unused to Clear
channel
Changing signalling on channel 23 from Unused to Clear
channel
Changing signalling on channel 24 from Unused to Clear
channel
Changing signalling on channel 25 from Unused to Clear
channel
Changing signalling on channel 26 from Unused to Clear
channel
Changing signalling on channel 27 from Unused to Clear
channel
Changing signalling on channel 28 from Unused to Clear
channel
Changing signalling on channel 29 from Unused to Clear
channel
Changing signalling on channel 30 from Unused to Clear
channel
Changing signalling on channel 31 from Unused to Clear
channel

I guess I'll have trouble getting all three cards to
work together on the same box.



 

[asterisk-users] Blacklist

[Nitesh Divecha]

Hello All,

I was wondering where does Asterisk stores the blacklist numbers?

I looked into the dialplan and it shows that it 
*"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB?


hyperion*CLI> show dialplan app-blacklist-add
[ Context 'app-blacklist-add' created by 'pbx_config' ]
 '1' =>1. *Set(DB(blacklist/${blacknr})=1)*
[pbx_config]
   2. Playback(num-was-successfully) 
[pbx_config]
   3. Playback(added)
[pbx_config]
   4. Wait(1)
[pbx_config]
   5. Hangup()   
[pbx_config]
 's' =>1. Answer()   
[pbx_config]
   2. Wait(1)
[pbx_config]
   3. Playback(enter-num-blacklist)  
[pbx_config]
   4. Set(TIMEOUT(digit)=5)  
[pbx_config]
   5. Set(TIMEOUT(response)=60)  
[pbx_config]
   6. Read(blacknr|then-press-pound) 
[pbx_config]
   7. SayDigits(${blacknr})  
[pbx_config]
   8. Playback(if-correct-press) 
[pbx_config]
   9. Playback(digits/1) 
[pbx_config]
[end]  10. Noop(Waiting for input)   
[pbx_config]
 Include =>'app-blacklist-add-custom'
[pbx_config]

hyperion*CLI>


Thanks,
Nitesh


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[asterisk-users] Save Key tone to MySQL DB

[Nitesh Divecha]

Hello All,

I am trying to build an IVR based survey system and I would like to save 
the key touch-tones to MySQL DB. It is a simple application where it 
determines the Caller ID and asks couple of questions and save the 
Callers input to the MySQL DB.


I am thinking of building IVR using FreePBX, but my main worry is how to 
save the Callers input?


Simple flow: -
1) Verify the CallerID, if not verified as for PIN number.
2) Once Caller is verified ask: -
   a) How would you rate our product? Press 1 for Excellent and Press 2 
for Bad.
   b) How did you like XYZ product? Press 1 for Excellent and Press 2 
for Bad.
   c) Are you satisfied with our product? Press 1 for Excellent and 
Press 2 for Bad.

3) End the call.

Now I want to save the Key input from the Caller for each question.

Hope these help...

Regards,
Nitesh


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Re: [asterisk-users] Re: Snom 320 voicemail key & MWI

[Nitesh Divecha]

Or you can specify "vmexten = *97" in sip.conf and your VM button will work.

Regards,
Nitesh








Nick Adams wrote:

Stephen Bosch wrote:

Ariel Monaco wrote:

Dear List,

I'm having a blinking MWI light on the snom 320 even when there's no
message waiting in Asterisk.
We've managed to make the voicemail button work using
fromdomain=192.168.0.1 in sip.conf
vmexten=2500 (our VoicemailMain application extension in
extensions.conf). We also added
notifymimetype=application/simple-message-summary also in sip.conf to
allow SIP simple MWI
notifications.

But the light is still blinking and there are no voicemail messages, 
any

ideas about how to address this
issue will be welcome.


You've mentioned how your Asterisk server is configured, but how is the
*phone* configured?

If the MWI light on the phone is set to use the wrong mailbox, you would
see a blinking light, even if you've erased all the messages in the
mailbox that is accessed from the voicemail button.

Two things are happening here:

1. You've got a button that you configure for retrieving messages
2. You've got a Message Waiting Indicator light that blinks when there
are messages in the specified mailbox.

Those are separate things -- you can have a button that retrieves from
one box and a light that indicates messages in another box.

Check your phone configuration again.


By default the Snom phones also use that light for missed calls.

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Re: [asterisk-users] TDM410P

[Nitesh Divecha]

Hello,

Here is my config: -

/etc/zaptel.conf

# T1 Configuration
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

span=3,1,0,esf,b8zs
bchan=49-71
dchan=72

span=4,1,0,esf,b8zs
bchan=73-95
dchan=96

/etc/asterisk/zapata-channels.conf >


; signalling = pri_cpe is USER
; signalling = pri_net is NETWORK

group = 1
switchtype = national
signalling = pri_net
context = from-zaptel
channel => 1-23

group = 2
switchtype = national
signalling = pri_net
context = from-zaptel
channel => 25-47

group = 3
switchtype = national
signalling = pri_net
context = from-zaptel
channel => 49-71

group = 4
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 73-95

I use FreePBX as my front-end to route calls... so I just assign the 
trunk groups which I want to use...


Regards,
Nitesh






Alexandre VERNIOL wrote:

HI all,

Does some one can give me his configuration (zapta.conf, zaptel.conf, 
sample for extensions.conf) for TDM410P card or similar (1/2/3/4 BRI 
card)


Thanks in advance.

Cheers,


Alex

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Re: [asterisk-users] GXV-3000 IP Video Phone

[Nitesh Divecha]

Hello,

So far yes... The Video phones are behaving good and all the 
functionality working.

I have 5 phone on the network and planning to put more by next week.

Cheers,
Nitesh




Noah Miller wrote:

Hi Nitesh -


Thanks everyone... The GXV-3000 IP Video Phone works with Asterisk 1.2
using H.263 Video Coder.

I had to update both phones firmware with new one...


Out of curiosity - do you like the phone?  I've looked for reviews,
but I haven't found any that rate the phone's functionality.


- Noah
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Re: [asterisk-users] GXV-3000 IP Video Phone

[Nitesh Divecha]

Thanks everyone... The GXV-3000 IP Video Phone works with Asterisk 1.2 
using H.263 Video Coder.


I had to update both phones firmware with new one...

Regards,
Nitesh










Jorge Mendoza wrote:

There are a patch for Asterisk 1.2 allowing h.264.
Please note as well that GXV-3000 last firmware works with H.263 too.

Jorge

Nitesh Divecha wrote:

Thanks Dave,

I did try with Asterisk 1.2 but it didn't work. The Video Phones came 
with H.264 Video Coder...


Regards,
Nitesh



Andreas van dem Helge wrote:

On 5/5/07, dave cantera <[EMAIL PROTECTED]> wrote:

nitesh,
you are correct.  you need 1.4.x...
daveC


It is supposed to have H.263, which does work with 1.2.x:


[general]
...
videosupport=yes
..

[video-enabled-sip-phone]
...
canreinvite=no
disallow=all
allow=ulaw
allow=h263
...
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Re: [asterisk-users] GXV-3000 IP Video Phone

[Nitesh Divecha]

Thanks Dave,

I did try with Asterisk 1.2 but it didn't work. The Video Phones came 
with H.264 Video Coder...


Regards,
Nitesh



Andreas van dem Helge wrote:

On 5/5/07, dave cantera <[EMAIL PROTECTED]> wrote:

nitesh,
you are correct.  you need 1.4.x...
daveC


It is supposed to have H.263, which does work with 1.2.x:


[general]
...
videosupport=yes
..

[video-enabled-sip-phone]
...
canreinvite=no
disallow=all
allow=ulaw
allow=h263
...
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[asterisk-users] GXV-3000 IP Video Phone

[Nitesh Divecha]

Hello All,

I just received some test units of Grandstream GXV-3000 IP Video Phone.

I did some research and looks like Asterisk 1.2 does not support video 
H.264 but Asterisk 1.4 does. Is it correct?


Actually I did try to test with Asterisk 1.2 and video did not 
initialize but voice worked...


Any advice?

Thanks,
Nitesh


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Re: [asterisk-users] T1/E1 Configuration

[Nitesh Divecha]

Thanks Everyone for the help...

Got the T1 UP and insvc with Cisco AS5350, but I am failing to send the 
call.
On the Cisco side I do not see any incoming call and on Asterisk side I 
get message saying "Channels unavailable", while all channels are available.

Can anyone post a working configuration for Asterisk T1 and Cisco conf?

Please... Thank you.

Regards,
Nitesh





Tzafrir Cohen wrote:

On Fri, May 04, 2007 at 02:28:32PM -0400, Andreas van dem Helge wrote:
  

On 5/4/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:


Thanks John,

How can I change my conf to NETWORK? Where can I find this information?
  

#signalling = pri_cpe
signalling = pri_net



nitpicking:

;signalling = pri_cpe
signalling = pri_net

(The comment character is ';' . '#' is reserved for special directives
of the sort of #include)

  


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Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04

[Nitesh Divecha]

Christian,

You can follow this procedure

http://www.aussievoip.com/wiki/freePBX-Ubuntu


Regards,
Nitesh









Christian wrote:

Hi,
I have already done:
apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from 
the latest sources.
So what should i do then? New to Ubuntu.
many thanks,
Christian


On 2007-05-04 at 17:00 Tzafrir Cohen wrote:

  

On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote:


Hi all,
Could someone please tell me how to make Asterisk start at boot on
  

Ubuntu Feisty 7.04?


Many thanks,
Christian

  

 apt-get install asterisk

Look at the init.d scripts.
Note that in Ubuntu, subdirectories under /var/run are deleted at boot,
and hence that script generates /var/run/asterisk (with proper
ownership) at boot time.

--
  Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] T1/E1 Configuration

[Nitesh Divecha]

Thanks John,

How can I change my conf to NETWORK? Where can I find this information?

Regards,
Nitesh



John Treble wrote:

Anyone can help me with this error?

May  4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error:
We think we're the CPE, but they think they're the CPE too.




Both ends of the T1 can't be running in CPE (USER) mode.  Typically, the
Telco is NETWORK and you are USER (CPE).  If you set your end of the T1 to
NETWORK mode I'll bet the D/B-channels will come up but **check with your
Telco first**. 



John Treble
Ottawa, Ontario, Canada


  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nitesh Divecha
Sent: May 4, 2007 10:43 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T1/E1 Configuration

Thanks Guys...

Got the T1/E1 Card working... Digium Engineers helped... According to
them TE405P card must load first and then the analog TDM400P.

Other thing which I messed up was that I changed the configuration to T1
but forgot to remove the Jumpers from the TE405P card. So that was
causing Asterisk to fail...

Its working now but can anyone clarify that... Do I have to remove all
four jumpers to make T1 card?

Anyone can help me with this error?

May  4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error:
We think we're the CPE, but they think they're the CPE too.


Regards,
Nitesh



Tzafrir Cohen wrote:


On Thu, May 03, 2007 at 11:03:59PM -0400, Nitesh Divecha wrote:

  

Hello All,

Can anyone please post their working T1/E1 configuration...

Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if
you run 'genzaptelconf' it created '/etc/asterisk/zapata-


channels.conf',


so please post that one also.



What do you have in /etc/asterisk/zapata.conf ?


  

Here is my configuration which is failing Asterisk to load... I have


two


cards TE405P and TDM400P: -



What error message do you get from asterisk at load time? You'll
typically see them in /var/log/asterisk/messages or
/var/log/asterisk/full


  

===
/etc/zaptel.conf
===
# T1 Configuration
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

span=3,1,0,esf,b8zs
bchan=49-71
dchan=72

span=4,1,0,esf,b8zs
bchan=73-95
dchan=96

fxsks=97
fxsks=98
fxsks=99
fxsks=100

# Global data

loadzone= us
defaultzone = us


/etc/asterisk/zapata-channels.conf

group = 1
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 1-23

group = 2
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 25-47

group = 3
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 49-71

group = 4
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 73-95

signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 97
; context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 98
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 99
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 100
context=default


Thanking in advance...

Cheers,
Nitesh


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Re: [asterisk-users] T1/E1 Configuration

[Nitesh Divecha]

Thanks Guys...

Got the T1/E1 Card working... Digium Engineers helped... According to 
them TE405P card must load first and then the analog TDM400P.


Other thing which I messed up was that I changed the configuration to T1 
but forgot to remove the Jumpers from the TE405P card. So that was 
causing Asterisk to fail...


Its working now but can anyone clarify that... Do I have to remove all 
four jumpers to make T1 card?


Anyone can help me with this error?

May  4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error: 
We think we're the CPE, but they think they're the CPE too.



Regards,
Nitesh



Tzafrir Cohen wrote:

On Thu, May 03, 2007 at 11:03:59PM -0400, Nitesh Divecha wrote:
  

Hello All,

Can anyone please post their working T1/E1 configuration...

Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if 
you run 'genzaptelconf' it created '/etc/asterisk/zapata-channels.conf', 
so please post that one also.



What do you have in /etc/asterisk/zapata.conf ?

  
Here is my configuration which is failing Asterisk to load... I have two 
cards TE405P and TDM400P: -



What error message do you get from asterisk at load time? You'll
typically see them in /var/log/asterisk/messages or
/var/log/asterisk/full

  

===
/etc/zaptel.conf
===
# T1 Configuration
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

span=3,1,0,esf,b8zs
bchan=49-71
dchan=72

span=4,1,0,esf,b8zs
bchan=73-95
dchan=96

fxsks=97
fxsks=98
fxsks=99
fxsks=100

# Global data

loadzone= us
defaultzone = us


/etc/asterisk/zapata-channels.conf

group = 1
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 1-23

group = 2
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 25-47

group = 3
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 49-71

group = 4
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 73-95

signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 97
; context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 98
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 99
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 100
context=default


Thanking in advance...

Cheers,
Nitesh


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[asterisk-users] T1/E1 Configuration

[Nitesh Divecha]

Hello All,

Can anyone please post their working T1/E1 configuration...

Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if 
you run 'genzaptelconf' it created '/etc/asterisk/zapata-channels.conf', 
so please post that one also.


Here is my configuration which is failing Asterisk to load... I have two 
cards TE405P and TDM400P: -

===
/etc/zaptel.conf
===
# T1 Configuration
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

span=3,1,0,esf,b8zs
bchan=49-71
dchan=72

span=4,1,0,esf,b8zs
bchan=73-95
dchan=96

fxsks=97
fxsks=98
fxsks=99
fxsks=100

# Global data

loadzone= us
defaultzone = us


/etc/asterisk/zapata-channels.conf

group = 1
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 1-23

group = 2
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 25-47

group = 3
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 49-71

group = 4
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 73-95

signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 97
; context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 98
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 99
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 100
context=default


Thanking in advance...

Cheers,
Nitesh


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Re: [asterisk-users] TDM400P and TE405P

[Nitesh Divecha]

Hello All,

To avoid conflicts I removed TE405P and left the TDM400P and 
reconfigured the card using "genzaptelconf".


When I run ztcfg -vv I saw the card and modules are loaded and also I 
used "ztmonitor 1 -v" and I saw the gain moving up and down. I did 
create trunks and outbound routes using FreePBX...


Now for some odd reason Asterisk is not picking up the incoming call 
from PSTN.


zapata-channels.conf
===
; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
;;; line="1 WCTDM/0/0"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 1
context=default
...


zapata.conf

#include zapata-channels.conf

Can anyone put some light why Asterisk is failing to pickup the call.


Regards,
Nitesh




Dave Miller wrote:

Nitesh Divecha wrote on 5/1/07 10:28 AM:

  

Is it possible to have both Digium cards installed on one Server
(TDM400P and TE405P)?

I have one site which requires both connection POT and T1/E1.

How can I configure both cards?



Should work just fine.  The Zaptel drivers will pick up both.  Just be
forewarned, the T1/E1 channels will all get numbered before the POTS
channels, no matter what order they're on the bus, so 1-24 will be your
T1 and 25-28 the POTS, for example.  (I think E1 goes to 32?)

  


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Re: [asterisk-users] TDM400P and TE405P

[Nitesh Divecha]

Dave Miller wrote:

Nitesh Divecha wrote on 5/1/07 10:28 AM:

  

Is it possible to have both Digium cards installed on one Server
(TDM400P and TE405P)?

I have one site which requires both connection POT and T1/E1.

How can I configure both cards?



Should work just fine.  The Zaptel drivers will pick up both.  Just be
forewarned, the T1/E1 channels will all get numbered before the POTS
channels, no matter what order they're on the bus, so 1-24 will be your
T1 and 25-28 the POTS, for example.  (I think E1 goes to 32?)

  

Thanks Dave,

From the Asterisk CLI when I do "zap show status", I get: -

hyperion*CLI> zap show status
Description  Alarms IRQ
bpviol CRC4 
Wildcard TDM400P REV I Board 1   OK 0  
0  0
T4XXP (PCI) Card 0 Span 1OK 0  
0  0
T4XXP (PCI) Card 0 Span 2OK 0  
0  0
T4XXP (PCI) Card 0 Span 3OK 0  
0  0
T4XXP (PCI) Card 0 Span 4OK 0  
0  0
hyperion*CLI>



Back of the cards I see four green lights on TDM400P, but no lights on 
TE405P... And right now I tried calling in on POT lines and Asterisk is 
not picking up the call...


Am I missing something here...?

Regards,
Nitesh




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