date:20070524

Re: [asterisk-users] problem with attended call transfer

2007-05-24 Thread Mandeep Singh Bhabha

Just add 
include => featuremap
in extensions.conf
i think this should help.


On Wed, May 23, 2007 at 12:59:39PM +, khawla khawla wrote:
> 
> 
> I am trying call transfer with asterisk. blind transfer (#) is working 
> perfectly, but attended transfer doesn't fonction (*2).
> I don't know what is the problem.
> Anyone could help?
> 
> _
> Lancez des recherches en toute s?curit? depuis n'importe quelle page Web. 
> T?l?chargez GRATUITEMENT Windows Live Toolbar aujourd'hui !
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-- 
С Уважением,
Мандип Сингх Бхабха
email: [EMAIL PROTECTED]

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RE: [asterisk-users] transfer call sip to zap

2007-05-24 Thread Alex Balashov


On Fri, 25 May 2007, Cosmin Prund wrote:

It just works. Simply transfer your call to the desired extension and 
let Asterisk take care of the details.


  Indeed.  A key appeal of Asterisk does lie precisely in that it 
abstracts, to a considerable degree, the chore of dealing with the

interoperation of channels of various media and signaling protocols.
It natively transcodes amongst interfaces and protocol stacks, so you
can take a call from a SIP peer and Dial(Zap/whatever...) in the dial
plan.

--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
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RE: [asterisk-users] transfer call sip to zap

2007-05-24 Thread Cosmin Prund

It just works. Simply transfer your call to the desired extension and
let Asterisk take care of the details.

 

--

Cosmin Prund

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of DiegoF
Sent: Friday, May 25, 2007 12:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] transfer call sip to zap

 

how to transfer a call from sip channel to zap channel

thanks

-- 
//  DiegoF  //

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[asterisk-users] Patton gateways (was: stream file not working but get data and exec background work)

2007-05-24 Thread Nick Seraphin

On Wed, 23 May 2007, Patrick Fortin wrote:

> Here is the setup
> 
> Pri SIP Gateway(Patton) - Asterisk (ztdummy) - SIP Phone

Wow... you're the first one I've seen who is using a Patton gateway, so I
hope you don't mind me asking you...  why did you go that route?

>From what I saw a while ago (I may be wrong) I thought Patton's SIP
gateways were way overpriced compared to other similar solutions.  Now I'm
going to have to go and double-check... maybe I misunderstood.

Or was there some sort of other reason why you chose Patton that made
price un-important?

Do you like their sip gateways?  What port density (or model?) do you
have?

-- Nick

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[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 114

2007-05-24 Thread JK


I am running asterisk 1.2.12.1

JK,

Message: 26
Date: Thu, 24 May 2007 21:40:31 -0700
From: JK <[EMAIL PROTECTED]>
Subject: [asterisk-users] Urgent: DTMF does not work with rtpmap:101
	telephone-event/8000 
To: asterisk-users@lists.digium.com

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Hello asterisk-users list.
I have been scratching my head for almost a week. We are trying to set a 
service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working.
In our scenario  the SP is sending call to our ser server and ser is 
forwarding the call to asterisk. In the asterisk debug I can see the 
DTMF keys are coming but ivr does not recognice those keys at all. I can 
see this in the debug. We are using ulaw and alaw for codec.


*May 24 20:14:00 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at 
XXX.XXX.XXX.XXX

May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at XXX.XXX.XXX.XXX
May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at 
XXX.XXX.XXX.XXX*



Voice part works great. I mean if I forward that call to asterisk sip 
user we can talk.
Every thing is working great with other SP. The only difference I can 
see is the rtpmap:101 telephone-event/8000.

With the working SP the rtpmap is rtpmap:100 telephone-event/8000.

I need solution ASAP, so please give me any hint you can.

Here is complete debug log for one call.

May 24 20:13:47 DEBUG[26583] chan_sip.c: Allocating new SIP dialog for 
9CB80474-

[EMAIL PROTECTED] - INVITE (With RTP)
May 24 20:13:47 DEBUG[26583] chan_sip.c:  Received INVITE (5) - 
Command in S

IP INVITE
May 24 20:13:47 DEBUG[26583] res_config_mysql.c: MySQL RealTime: 
Everything is f

ine.
May 24 20:13:47 DEBUG[26583] res_config_mysql.c: MySQL RealTime: 
Retrieve SQL: S

ELECT * FROM cc_sip_buddies WHERE name = '6056911122'
May 24 20:13:47 DEBUG[26583] chan_sip.c: Setting NAT on RTP to 0
May 24 20:13:47 DEBUG[26583] chan_sip.c: Checking SIP call limits for device
May 24 20:13:47 DEBUG[26583] chan_sip.c: Updating call counter for 
incoming call
May 24 20:13:47 DEBUG[26583] chan_sip.c: build_route: Record-Route hop: 

.216.2.212;ftag=E3F05090-621;lr=on>
May 24 20:13:47 DEBUG[26583] chan_sip.c: build_route: Record-Route hop: 

May 24 20:13:47 DEBUG[26583] chan_sip.c: build_route: Record-Route hop: 

May 24 20:13:47 DEBUG[26574] chan_sip.c: Checking device state for peer 
XXX.XXX..XXX

May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'NoOp'
May 24 20:13:47 DEBUG[26574] devicestate.c: Changing state for 
SIP/XXX.XXX.XXX.XXX -

state 2 (In use)
May 24 20:13:47 DEBUG[26804] app_queue.c: Device 'SIP/XXX.XXX.XXX.XXX' 
changed to st
ate '2' (In use) but we don't care because they're not a member of any 
queue.

May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Goto'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Goto'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Answer'
May 24 20:13:47 DEBUG[26803] chan_sip.c: 
sip_answer(SIP/XXX.XXX.XXX.XXX-b7b03730)
May 24 20:13:47 DEBUG[26574] chan_sip.c: Checking device state for peer 
XXX.XXX..XXX
May 24 20:13:47 DEBUG[26574] channel.c: Avoiding initial deadlock for 
'SIP/XXX.XXX..XXX-b7b03730'

May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Wait'
May 24 20:13:47 DEBUG[26574] devicestate.c: Changing state for 
SIP/XXX.XXX.XXX.XXX -

state 2 (In use)
May 24 20:13:47 DEBUG[26805] app_queue.c: Device 'SIP/XXX.XXX.XXX.XXX' 
changed to st
ate '2' (In use) but we don't care because they're not a member of any 
queue.
May 24 20:13:47 DEBUG[26583] chan_sip.c: = Found Their Call ID: 
9CB80474-98311DC
[EMAIL PROTECTED] Their Tag E3F05090-621 Our tag: 
as3becafbb
May 24 20:13:47 DEBUG[26583] chan_sip.c:  Received ACK (6) - Command 
in SIP

ACK
May 24 20:13:47 DEBUG[26583] chan_sip.c: Stopping retransmission on 
'9CB80474-98

[EMAIL PROTECTED]' of Response 101: Match Found
May 24 20:13:48 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:48 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:48 DEBUG[26803] pbx.c: Launching 'BackGround'
May 24 20:13:48 DEBUG[26803] channel.c: Set channel 
SIP/XXX.XXX.XXX.XXX-b7b03730 to

write format slin
May 24 20:13:48 DEBUG[26803] rtp.c: Ooh, format changed from unknown to ulaw
May 24 20:13:48 DEBUG[26803] channel.c: Scheduling timer at 160 sample 
intervals

May 24 20:13:49 DEBUG[26803] rtp.c: Got RTCP report of 132 bytes
May 24 20:13:52 DEBUG[26803] rtp.c: Got RTCP report of 132 bytes
May 24 20:13:53 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at XXX.XXX.XXX.XXX
May 24 20:13:58 DEBUG[26803] rtp.c: Got RTCP report of 132 bytes
May 24 20:14:00 DEBUG[26803] channel.c: Scheduling timer at 58 sample 
intervals
May 24 20:14:00 DEBUG[26803] channel.c: Scheduling timer at 0 sample 
intervals
May 24 20:14:00 DEBUG[26803] channel.c: Scheduling timer at 0 sample 
intervals
May 24 20:14:00 DEBUG[26803] channel.c: Set channel 
SIP/XXX.

Re: [asterisk-users] GUI: Not Found. Move along

2007-05-24 Thread Russell Bryant


Tim Verscheure wrote:

yes!!

2007/5/21, Guilherme Góes <[EMAIL PROTECTED]>:

Did you acess the page at port 8088 ? I.e.: http://192.168.0.1:8088


http://192.168.0.1:8088/asterisk/static/config/cfgbasic.html

--
Russell Bryant
Software Engineer
Digium, Inc.
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Re: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000

2007-05-24 Thread Alex Balashov



JK,

In-band or RFC2833 DTMF signaling?

Also, unless you have SER configured with a media proxy, the actual "call" 
is not running through SER.  It's a signaling proxy only.


--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
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[asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000

2007-05-24 Thread JK


Hello asterisk-users list.
I have been scratching my head for almost a week. We are trying to set a 
service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working.
In our scenario  the SP is sending call to our ser server and ser is 
forwarding the call to asterisk. In the asterisk debug I can see the 
DTMF keys are coming but ivr does not recognice those keys at all. I can 
see this in the debug. We are using ulaw and alaw for codec.


*May 24 20:14:00 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at 
XXX.XXX.XXX.XXX

May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at XXX.XXX.XXX.XXX
May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at 
XXX.XXX.XXX.XXX*



Voice part works great. I mean if I forward that call to asterisk sip 
user we can talk.
Every thing is working great with other SP. The only difference I can 
see is the rtpmap:101 telephone-event/8000.

With the working SP the rtpmap is rtpmap:100 telephone-event/8000.

I need solution ASAP, so please give me any hint you can.

Here is complete debug log for one call.

May 24 20:13:47 DEBUG[26583] chan_sip.c: Allocating new SIP dialog for 
9CB80474-

[EMAIL PROTECTED] - INVITE (With RTP)
May 24 20:13:47 DEBUG[26583] chan_sip.c:  Received INVITE (5) - 
Command in S

IP INVITE
May 24 20:13:47 DEBUG[26583] res_config_mysql.c: MySQL RealTime: 
Everything is f

ine.
May 24 20:13:47 DEBUG[26583] res_config_mysql.c: MySQL RealTime: 
Retrieve SQL: S

ELECT * FROM cc_sip_buddies WHERE name = '6056911122'
May 24 20:13:47 DEBUG[26583] chan_sip.c: Setting NAT on RTP to 0
May 24 20:13:47 DEBUG[26583] chan_sip.c: Checking SIP call limits for device
May 24 20:13:47 DEBUG[26583] chan_sip.c: Updating call counter for 
incoming call
May 24 20:13:47 DEBUG[26583] chan_sip.c: build_route: Record-Route hop: 

.216.2.212;ftag=E3F05090-621;lr=on>
May 24 20:13:47 DEBUG[26583] chan_sip.c: build_route: Record-Route hop: 

May 24 20:13:47 DEBUG[26583] chan_sip.c: build_route: Record-Route hop: 

May 24 20:13:47 DEBUG[26574] chan_sip.c: Checking device state for peer 
XXX.XXX..XXX

May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'NoOp'
May 24 20:13:47 DEBUG[26574] devicestate.c: Changing state for 
SIP/XXX.XXX.XXX.XXX -

state 2 (In use)
May 24 20:13:47 DEBUG[26804] app_queue.c: Device 'SIP/XXX.XXX.XXX.XXX' 
changed to st
ate '2' (In use) but we don't care because they're not a member of any 
queue.

May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Goto'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Goto'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Answer'
May 24 20:13:47 DEBUG[26803] chan_sip.c: 
sip_answer(SIP/XXX.XXX.XXX.XXX-b7b03730)
May 24 20:13:47 DEBUG[26574] chan_sip.c: Checking device state for peer 
XXX.XXX..XXX
May 24 20:13:47 DEBUG[26574] channel.c: Avoiding initial deadlock for 
'SIP/XXX.XXX..XXX-b7b03730'

May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Wait'
May 24 20:13:47 DEBUG[26574] devicestate.c: Changing state for 
SIP/XXX.XXX.XXX.XXX -

state 2 (In use)
May 24 20:13:47 DEBUG[26805] app_queue.c: Device 'SIP/XXX.XXX.XXX.XXX' 
changed to st
ate '2' (In use) but we don't care because they're not a member of any 
queue.
May 24 20:13:47 DEBUG[26583] chan_sip.c: = Found Their Call ID: 
9CB80474-98311DC
[EMAIL PROTECTED] Their Tag E3F05090-621 Our tag: 
as3becafbb
May 24 20:13:47 DEBUG[26583] chan_sip.c:  Received ACK (6) - Command 
in SIP

ACK
May 24 20:13:47 DEBUG[26583] chan_sip.c: Stopping retransmission on 
'9CB80474-98

[EMAIL PROTECTED]' of Response 101: Match Found
May 24 20:13:48 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:48 DEBUG[26803] pbx.c: Launching 'Set'
May 24 20:13:48 DEBUG[26803] pbx.c: Launching 'BackGround'
May 24 20:13:48 DEBUG[26803] channel.c: Set channel 
SIP/XXX.XXX.XXX.XXX-b7b03730 to

write format slin
May 24 20:13:48 DEBUG[26803] rtp.c: Ooh, format changed from unknown to ulaw
May 24 20:13:48 DEBUG[26803] channel.c: Scheduling timer at 160 sample 
intervals

May 24 20:13:49 DEBUG[26803] rtp.c: Got RTCP report of 132 bytes
May 24 20:13:52 DEBUG[26803] rtp.c: Got RTCP report of 132 bytes
May 24 20:13:53 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at XXX.XXX.XXX.XXX
May 24 20:13:58 DEBUG[26803] rtp.c: Got RTCP report of 132 bytes
May 24 20:14:00 DEBUG[26803] channel.c: Scheduling timer at 58 sample 
intervals
May 24 20:14:00 DEBUG[26803] channel.c: Scheduling timer at 0 sample 
intervals
May 24 20:14:00 DEBUG[26803] channel.c: Scheduling timer at 0 sample 
intervals
May 24 20:14:00 DEBUG[26803] channel.c: Set channel 
SIP/XXX.XXX.XXX.XXX-b7b03730 to

write format ulaw
May 24 20:14:00 DEBUG[26803] res_config_mysql.c: MySQL RealTime: 
Everything is f

ine.
May 24 20:14:00 DEBUG[26803] res_config_mysql.c: MySQL RealTime: 
Retrieve SQL: S
ELECT * FROM extensions WHERE exten = 's' AND context = 'ext-local' AND 
priority

= '8'
May 24 20:14:00 DEBUG[2

[asterisk-users] Echo on hard SIP devices...

2007-05-24 Thread Carlos Chavez

 We have an installation with around 50 sip phones but only 5 of those are
hardware.  There are three Grandstream phones, one Snom and one PAP2T.  We are
running Asterisk 1.2.8 with an E1 (R2).  Only the hard phones are having
problems which are either echo or distortion.  The softphones all work fine
and no one is reporting any problems.

 They are using 3Com switches which are fairly new.  I have really tried
all the settings I can think of and it seems impossible that all 5 hard phones
are defective.  Obviously the customer is irritated because the hard phones
belong to the director and the receptionist and they notice the problem all
day.  I still think the problem may be with the switch but I just want to
check if anyone has had an experience like this before.

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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Re: [asterisk-users] TE205P, E1, Panasonic PBX and hang-up issues

2007-05-24 Thread C F


Are you sure the panasonic is TVP 100? I have installed over 50
Panasonic systems in my life, and service many more, I have never
heard of that system, and a quick google shows it's just a VoiceMail
system and not a PBX.

On 5/23/07, Barry O'Donovan <[EMAIL PROTECTED]> wrote:


Hey folks,

I have a Digium TE205P working as a man in the middle:

PRI line  Asterisk/TE205P  PBX

The PBX is a Panasonic KX - TVP 100.

Everything is working great except for one little issue. Asterisk isn't
hanging up the PRI B channel when the PBX channel is hung up.

I don't want to overload you with information but please ask if more is
needed. I suspect I'm really hoping someone who had a similar problem with
just say "ah, I know what that is!".

Versions in use for Zaptel, LibPRI and Asterisk are all the SVN 1.4 branch.

To replicate:

1. dial a mobile (say) from one of the PBX phones;
2. when you here a ring tone, hang up the PBX phone;
3. the mobile continues to ring.

The verbose output is:

-- Accepting overlap call from '' to '' on channel 0/17, span
2
-- Starting simple switch on 'Zap/48-1'
-- Executing [EMAIL PROTECTED]:1]
Set("Zap/48-1", "RECORDFILE=/srv/recordings/live/1179858572.0") in new stack
-- Executing [EMAIL PROTECTED]:2]
MixMonitor("Zap/48-1", "/srv/recordings/live/1179858572.0.wav|b") in new
stack
-- Executing [EMAIL PROTECTED]:3] SetCallerPres("Zap/48-1", "allowed") in
new stack
-- Executing [EMAIL PROTECTED]:4] SetCallerID("Zap/48-1", "5400") in new
stack
-- Executing [EMAIL PROTECTED]:5] Dial("Zap/48-1", "Zap/g0/0868017669") in
new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0868017669
  == Begin MixMonitor Recording Zap/48-1
-- Zap/1-1 is ringing
-- Channel 0/17, span 2 got hangup request, cause 16
-- Zap/1-1 answered Zap/48-1
-- Channel 0/1, span 1 got hangup request, cause 0
-- Hungup 'Zap/1-1'
  == Spawn extension (pbx, 0868017669, 5) exited non-zero on 'Zap/48-1'
  == End MixMonitor Recording Zap/48-1
-- Hungup 'Zap/48-1'
asterisk1*CLI>


Any suggestions or fixes that you might have from prior instances would be
greatly appreciated.

Thanks a million,

Barry O'Donovan
http://www.barryodonovan.com/

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Re: [asterisk-users] CDR on channel 'IAX2/u92613106-3' already started

2007-05-24 Thread Mike Diehl

On Thursday 24 May 2007 06:35, Steve Murphy wrote:
> On Wed, 2007-05-23 at 20:51 -0600, Mike Diehl wrote:
> > Hi all,
> >
> > I'm having a problem with an asterisk server being unable to call certain
> > cellphones and answering machines.  Anytime the person answers the phone
> > call, everything works well.  But when the call goes to voicemail or an
> > answering machine, I get the error message below:
> > --- snip ---
>
> Mike--
>
> I assume you're using 1.4 or trunk; normally this shouldn't be a big
> deal. Exactly which version of 1.4 are you using? Have you tried the
> latest SVN version? I made some fixes concerning check_start.

I'm using 1.4.4.  The really strange thing is that the call FAILS when an 
answering machine or voicemail picks up.  Recalling the same number but 
having a person pick up works like a charm.

Any ideas?

> murf

-- 
Mike Diehl
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Re: [asterisk-users] Nokia release

2007-05-24 Thread Mark Phillips

Nokia N95 available via ATT/Cingular for $795 with a 2 year contract. It
was advertised in the New Jersey Star Ledger this morning.

Mark

On Thu, 2007-05-24 at 18:42 +0500, Rizwan Hisham wrote:
> Hi all,
> sorry to ask you something not related to asterisk, but i really want
> to know whether the Nokia N95 cell phone is released in the USA or
> not? if somebody from USA knows, plz reply.
> 
> -- 
> Rizwan Hisham 
> Software Engineer
> AXVOICE Inc. 
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Re: [asterisk-users] vmoutcall]

2007-05-24 Thread Doug Lytle

--> Perhaps someone can share how? 

First you need to give them the option of turning the feature on and 
off.  I do it with the following:


[callback-activate]

; ***
; Callback activate/deactivate.  If this function
; is enabled and there is a call file in the form
; of ${EXTEN}.call, then Asterisk will call the
; phone number contained within the .call file 150
; seconds after a voicemail has been left.
; ***

exten => 80*,1,Set(CALLBACK=${DB(vmcallback/${CALLERIDNUM})})
exten => 80*,2,GotoIf($["${CALLBACK}" = "YES"]?80*,3:80*,101)
exten => 80*,3,Set(DB(vmcallback/${CALLERIDNUM})=NO)
exten => 80*,4,Playback(local/stutter)
exten => 80*,5,Playback(de-activated)
exten => 80*,6,Hangup()
exten => 80*,101,Set(DB(vmcallback/${CALLERIDNUM})=YES)
exten => 80*,102,Playback(local/stutter)
exten => 80*,103,Playback(activated)
exten => 80*,104,Hangup()

Then you need to do a database look up every place in your dial plan 
where voice mail may be left, I do it as such:


[macro-sip.extensions]

exten => s,1,Set(CALLBACK=${DB(vmcallback/${ARG1})})
exten => s,n,SetMusicOnHold(cd)
exten => s,n,Dial(SIP/${ARG1},28,tWw)
exten => s,n,NoOP(Dial Status: ${DIALSTATUS})
exten => s,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,GotoIf($["${CALLBACK}" = 
"YES"]?s-NOANSWER,2:s-NOANSWER,3)

exten => s-NOANSWER,2,System(/usr/local/bin/vm-callout.sh ${ARG1})
exten => s-NOANSWER,3,Voicemail([EMAIL PROTECTED])

If CALLBACK=YES, then run the script that copies the call file into the 
outgoing directory.  It uses touch to set the date on the file 150 
seconds into the future.  This prevents the system calling the user 
while voice mail is still being left.


The call file links into the dial plan that loops the message 4 times 
waiting for acknowledgment by pressing 1 to collect voice mail.


[voice-mail-callback]

; 
; Set timeouts
; 

exten => s,1,Set(TIMEOUT(response)=6)
exten => s,2,Set(TIMEOUT(digit)=3)
exten => s,3,Wait(1)
exten => s,4,Set(COUNT=0)

; ***
; Play, your attention is required, press 1 to
; collect voice mail
; ***

exten => s,5,Background(attention-required)
exten => s,6,Background(press-1)
exten => s,7,Background(to-collect-voicemail)

; *
; If 1 is pressed, then play transfer and
; then jump to voice-mail context.
; *

exten => 1,1,Playback(pbx-transfer)
exten => 1,2,Goto(voice-mail,s,1)

; 
; Setup a variable to count the number of
; times the message has been played, when
; $COUNT reaches > 3, play you've taken
; to long to dial and hangup.
; 

exten => t,1,Set(COUNT=$[${COUNT} + 1])
exten => t,2,NoOP(${COUNT})
exten => t,3,GotoIf($[ ${COUNT} > 3 ]?103)
exten => t,4,Goto(voice-mail-callback,s,5)
exten => t,103,Playback(local/tolong-todial)
exten => t,104,Playback(goodbye)
exten => t,105,Hangup()

exten => i,1,Playback(local/sorry-invalid-choice)
exten => i,2,Set(COUNT=$[${COUNT} + 1])
exten => i,3,NoOP(${COUNT})
exten => i,4,Goto(voice-mail-callback,s,5)

exten => h,1,NoOP(Hungup)


Doug




--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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RE: [asterisk-users] Call Center Application

2007-05-24 Thread Senad Jordanovic

bilal ghayyad wrote:
> Hi list;
> 
> I am looking for an application that can be used with call center, in
> this application we can integrate the telephony part of the call
> center (like CTI Client ad so on), any one can advise for a good
> application to be used with Asterisk Call Center?   
> 
> - Note: The application to be customized easy, to be able to use it
> with Banking, Telecom, Oil, ..  etc. 
> 
> Regards
> Bilal

Try PBXware call centre edition. Full call centre stats, real time
monitoring, unlimited agents etc.


http://www.bicomsystems.com/products/C/P/319/154_2573/


Regards,


Senad
www.bicomsystems.com




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[asterisk-users] Call Center Application

2007-05-24 Thread bilal ghayyad

Hi list;

I am looking for an application that can be used with
call center, in this application we can integrate the
telephony part of the call center (like CTI Client ad
so on), any one can advise for a good application to
be used with Asterisk Call Center?

- Note: The application to be customized easy, to be
able to use it with Banking, Telecom, Oil, ..  etc.

Regards
Bilal


   
Get
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Re: [asterisk-users] Cisco CP-7970G

2007-05-24 Thread lists

On Thu, May 24, 2007 6:27 pm, Anthony Francis <[EMAIL PROTECTED]> said:

> [EMAIL PROTECTED] wrote:
>> Hi all,
>>
>> I just bought the 7970G phone. It's a beautiful phone. In trying to make it 
>> work
>> with Asterisk, I've read many posts on the net. However, all of them make
>> reference to having to install the SIP firmware on the phone. Where can I get
>> it?
>>
>> Thanks
>>
>>
>>
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>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> Download it from cisco www.cisco.com
> 

Just tried that. It seems that you need a Cisco Service Agreement before you 
can download it. Is that correct? Is that crazy?

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Re: [asterisk-users] Conference room as Music on Hold

2007-05-24 Thread Kyle Hagan

See: 
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf


Look for: Using a sound card as the source.

You can then plug in a radio or what ever.



shadowym wrote:
 
Here is what I am trying to do.  I have a SIP soft phone running on a PC

that is streaming a local radio station.  I assigned "mono out" in XP
Equalizer as the mic so now I have the softphone streaming audio.  I then
create a conference room and dial that room from the softphone.  Now anyone
who joins the conference room hears the streaming audio.

How can I configure Asterisk so that when musiconhold is invoked it
automatically joins that conference room?

The reason I want to do this is because the radio station uses windows media
and I haven't been able to get it to work directly on Asterisk.  Even after
following the mplayer instructions on the Asterisk wiki.

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Re: [asterisk-users] Integrated T1

2007-05-24 Thread BJ Weschke

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

On Thu, 24 May 2007, William Moore wrote:

> On 5/24/07, Jeremy Mann <[EMAIL PROTECTED]> wrote:
>> Can an asterisk box equipped with a Digium T1 card handle Integrated T1
>> circuits?  I have a T1 with 768k data and the remaining channels voice, can
>> the asterisk box do the Data routing + Voice processing?
>
> Yes, zaptel will create a device node for you.  Take a look at the
> set-hdlc tool in zaptel and the less common channel types in the
> default zaptel config file (rawhdlc is one, there are also others).

   Interesting!  Is this a relatively recent development?

   I stand corrected.

   Is there a way to make PPP-encapsulated T1 work as well?

Yes. I've got a client with a TE212P with a PRI channelized on one
port and a PPP encapsulated IP T1 on the other. It works fine.

BJ

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] Conference room as Music on Hold

2007-05-24 Thread Alex Balashov


On Thu, 24 May 2007, shadowym wrote:


How can I configure Asterisk so that when musiconhold is invoked it
automatically joins that conference room?


  Can't say for sure, but the best intuition would be to try to look at
how your phone's "hold" button implements its function on the SIP layer
and see if you can build something into the dial plan and/or features.conf
that would have the effect of placing the user into a MeetMe room.

The reason I want to do this is because the radio station uses windows 
media and I haven't been able to get it to work directly on Asterisk. 
Even after following the mplayer instructions on the Asterisk wiki.


  Why doesn't it work?  Live transcoding via mplayer should work fine.

  That said, it can be a rather CPU-intensive process and is rather
pointless.  By the time the audio is butchered by that process +
some load, not to mention the underlying nature of a 64kbps 3.1 KHz
speech bearer capability, it won't sound nearly as good as conventional
radio-on-hold most likely.

  Another thing you could try is hook mpg123 into the picture, instead
of using native MP3 decoding.  Set up the WMA to be streamed as MP3 on
some Shoutcast server or something, then feed mpg123 a file that actually 
contains a reference to an MP3 stream URI.  See how that works for you.


--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
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Re: [asterisk-users] Cisco CP-7970G

2007-05-24 Thread Anthony Francis


[EMAIL PROTECTED] wrote:

Hi all,

I just bought the 7970G phone. It's a beautiful phone. In trying to make it 
work with Asterisk, I've read many posts on the net. However, all of them make 
reference to having to install the SIP firmware on the phone. Where can I get 
it?

Thanks



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Download it from cisco www.cisco.com
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RE: [asterisk-users] vmoutcall

2007-05-24 Thread Paul Aviles

Perhaps someone can share how? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, May 24, 2007 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] vmoutcall

Paul Aviles wrote:
> Hello guys,
>
> I have been looking for a way to call a cell phone after someone has 
> left a
>   
This can easily be done with database lookups and .call files

> to accomplishing this? Most analog pbx's have this feature and I am 
> amazed Asterisk does not natively.
>   

It can be done natively; within the dial plan.

Doug

-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."


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[asterisk-users] Login log out support

2007-05-24 Thread Paul Aviles

is there a way to support login and logout functionality in a phone? We are
using Cisco 7940 and 7960 phones and have 2 shift. We want to be able to use
the same phone using like 2 different extensions. The phone will then
"remember" your settings if possible, if anyone has left you a voice mail
etc.
 
Is this possible?
 
Regards,
 
Paul
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[asterisk-users] Cisco CP-7970G

2007-05-24 Thread lists

Hi all,

I just bought the 7970G phone. It's a beautiful phone. In trying to make it 
work with Asterisk, I've read many posts on the net. However, all of them make 
reference to having to install the SIP firmware on the phone. Where can I get 
it?

Thanks



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RE: [asterisk-users] Integrated T1

2007-05-24 Thread Steve Totaro

Both Digium and Sangoma support this configuration but something tells
me that Sangoma will be easier to setup.  That something is trying it
several years ago when there were little bits and pieces on how to do it
spread all over.  I never did get it to work.  Digium listed it as a
feature but when called for support, they said it was not a "supported"
feature.

You may have better luck these days.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com 
KB3OPB

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: Thursday, May 24, 2007 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Integrated T1

On Thu, 24 May 2007, Jeremy Mann wrote:

> Can an asterisk box equipped with a Digium T1 card handle
Integrated T1

> circuits?  I have a T1 with 768k data and the remaining
channels voice,

> can the asterisk box do the Data routing + Voice processing?

I'm not certain, but I believe the Sangoma WANrouter/WANPipe cards are
capable of this. Call Sangoma and ask them if it is possible.

Tom

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[asterisk-users] transfer call sip to zap

2007-05-24 Thread DiegoF


how to transfer a call from sip channel to zap channel

thanks

--
//  DiegoF  //
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[asterisk-users] Re: Integrated T1

2007-05-24 Thread David Cook

On Thu, 24 May 2007, Jeremy Mann wrote:

>> Can an asterisk box equipped with a Digium T1 card handle Integrated
T1
>> circuits?  I have a T1 with 768k data and the remaining channels
voice,
>> can the asterisk box do the Data routing + Voice processing?

>   The Zaptel/Asterisk infrastructure can definitely break particular
>timeslots out of the T1 for voice, but it is not my impression that
>any existing WAN drivers for Linux support Digium cards or cohabitation
>with Zapata and can give you a serial data interface on other channels.

There are obvious risk factors with the scenario of your Asterisk box
being your CSU/DSU/Firewall & Router but for a small office this can
actually be a good thing.

Sangoma cards with their Wanpipe drivers can do this for you.

dbc.
--
David Cook


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[asterisk-users] Conference room as Music on Hold

2007-05-24 Thread shadowym


 
Here is what I am trying to do.  I have a SIP soft phone running on a PC
that is streaming a local radio station.  I assigned "mono out" in XP
Equalizer as the mic so now I have the softphone streaming audio.  I then
create a conference room and dial that room from the softphone.  Now anyone
who joins the conference room hears the streaming audio.

How can I configure Asterisk so that when musiconhold is invoked it
automatically joins that conference room?

The reason I want to do this is because the radio station uses windows media
and I haven't been able to get it to work directly on Asterisk.  Even after
following the mplayer instructions on the Asterisk wiki.

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Re: [asterisk-users] vmoutcall

2007-05-24 Thread Doug Lytle


Paul Aviles wrote:

Hello guys,

I have been looking for a way to call a cell phone after someone has left a
  

This can easily be done with database lookups and .call files


to accomplishing this? Most analog pbx's have this feature and I am amazed
Asterisk does not natively.
  


It can be done natively; within the dial plan.

Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] Bottom line on fax reception

2007-05-24 Thread Alex Balashov


On Thu, 24 May 2007, shadowym wrote:

So what is the bottom line?  Does it work or not.  I've heard stories it 
works, it doesn't work, it kinda sorta works when it's not raining out 
side. Everything under the rainbow.


  If you're talking about running it as analog pass-thru over G.711u over 
SIP, especially over the Internet-at-large, that's the answer;  it mostly 
works, unless it doesn't.  Analog fax is highly susceptible to even very 
minute distortions introduced by jitter, 1-2% packet loss, and/or 
out-of-sequence RTP payloads.  That's not something software can really

fix, except to the extent that T.38 may do so.

  If you're talking about picking faxes up over TDM/POTS, it should work 
fine. I've set it up a number of times and never had any problems.


--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
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Re: [asterisk-users] Bottom line on fax reception

2007-05-24 Thread Doug Lytle


shadowym wrote:
 
So what is the bottom line?  Does it work or not.  I've heard stories it
  


As it has been said many many times before, Fax detection is an art and 
most of the time is not reliable.  Faxing on the other hand, using 
iaxmodem along with HylaFAX+ works very well.  Search the archives.


Doug


--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] Bottom line on fax reception

2007-05-24 Thread mail-lists


shadowym wrote:
 
So what is the bottom line?  Does it work or not.  I've heard stories it

works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.

What's the bottom line with recent updates on 1.2.x?  Is it production ready
for fax?  By production ready I mean that it just works all the time and
doesn't need any babysitting.  Do I have to worry about dropped lines,
sometimes not detecting incoming fax toneyada yada.  


One simple question - VOIP or PSTN?
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Re: [asterisk-users] Integrated T1

2007-05-24 Thread Tom Rymes


On Thu, 24 May 2007, Jeremy Mann wrote:

> Can an asterisk box equipped with a Digium T1 card handle  
Integrated T1
> circuits?  I have a T1 with 768k data and the remaining channels  
voice,

> can the asterisk box do the Data routing + Voice processing?


I'm not certain, but I believe the Sangoma WANrouter/WANPipe cards  
are capable of this. Call Sangoma and ask them if it is possible.


Tom

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Re: [asterisk-users] Integrated T1

2007-05-24 Thread Jon Pounder


Quoting Alex Balashov <[EMAIL PROTECTED]>:


On Thu, 24 May 2007, William Moore wrote:


On 5/24/07, Jeremy Mann <[EMAIL PROTECTED]> wrote:

Can an asterisk box equipped with a Digium T1 card handle Integrated T1
circuits?  I have a T1 with 768k data and the remaining channels voice, can
the asterisk box do the Data routing + Voice processing?


Yes, zaptel will create a device node for you.  Take a look at the
set-hdlc tool in zaptel and the less common channel types in the
default zaptel config file (rawhdlc is one, there are also others).


  Interesting!  Is this a relatively recent development?


its one of the oldest things related to asterisk in general.



  I stand corrected.

  Is there a way to make PPP-encapsulated T1 work as well?

--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
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Jon Pounder

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_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread 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] 
> 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] basic 3+ way conference call on plain old phones

2007-05-24 Thread pedro noticioso

hi guys, is it possible to do a basic 3-or-more-way
conference call when the phones dont support it? I am
fully aware of this concept on expensive phones like
this one:

Grandstream GXP 2000 -Conference call 3-way
http://www.youtube.com/watch?v=hlZ6JqE1MT4

The problem is that the basic plain old commercial PBX
supports 3-way calling in ugly old phones like this
one:

http://www.neo-shop.com/tiendas/0009/varios/telefono%20TEIDE-1.jpg

connected to an ata like this one:

http://www.egk.com.ar/imagenes/hardware/sipura2.jpg

The idea is to be caller (A): dial calle (B), once (B)
answers press on HOOK or something else to send them
to MOH, then dial callee (C), talk to him a little
too, then press the same HOOK or something else and
the 3, (A)(B) and (C) in a conference call.

Unlike the grandstream, this would definitelly have to
be done by *, isnt this part of the basic
functionality like voicemail that is already done and
a couple lines in the config files it will work on all
phones done by *?

if not, then, how do you recommend me to it? 

the closest I have seen to shat I am looking for is

http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro

is there a better alternative?

any thoughts?

thanks a lot!



   
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 a little couch potato? 
Check out fun summer activities for kids.
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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread Anthony Francis

Something like this would be fairly easy to write, but it will cost you. :)

Nitesh Divecha wrote:

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] 
> 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]

> >> 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] Basic connection between Mitel 3300 ICP and Asterisk (trixbox) - from a clueless newbie....

2007-05-24 Thread Alex Crow

Hi all,

Our company has deployed a Mitel 3300 system (only about 2.5 years ago)
and we are experimenting with setting up Asterisk in our head office
(for business continuity, ie we have a bird flu epidemic and no-one can
come in, therefore use SIP softphones at home to co-ordinate activity)
and at a remote site in the Isle of Man (connected via 2Mbps SDSL)

Ideally we'd like anyone on either Asterisk servers (IOM and London) to
be able to dial anyone internally on the Mitel 3300 and vice-versa. We
have got *one* SIP license so far for the Mitel for testing purposes.

I am a bit crap on telephony, but as I have gathered so far we should be
able to connect the two systems via either QSIG (with an appropriate
card on the Asterisk server), DPNSS (which I'm not sure if any Asterisk
compatible hardware supports) or SIP (I'm happy setting up clients, but
have no clue with inter-PBX stuff).

I don't really care about any special features as long as the Mitel
numbers can call SIP users in London or IOM and the other way round.

I am planning to get at least 1 BRI pulled into the IOM office for PSTN
access, btw.

Any help you can offer would be gratefully received.

Cheers

Alex

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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread kido


Hi,

I think is it easily doable with an AGI script, but i am not sure if 
there is any builtin function to do it.

So you might want to look into that (AGI script)

Kido

Nitesh Divecha a écrit :

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] 
> 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

2007-05-24 Thread 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] 
> 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]

> >> 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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Nortex Networks

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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread 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] meetme sounds

2007-05-24 Thread Andrew Kohlsmith

On Thursday 24 May 2007 11:30 am, Steve Edwards wrote:
> As I remember, the "key" was to add code to conf_run() to take the user
> out of the conference, play the custom sound file, and put them back into
> the conference. These in/out steps are needed to keep that user in sync
> with the conference. Otherwise, their audio will be offset by the length
> of the sound file.

Eep; I wonder if it would have been easier to mix the sound in to just their 
copy of the conference... if people were entering or leaving when others were 
talking it would create "holes" in the creator's audio.

-A.
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Re: [asterisk-users] SCCP

2007-05-24 Thread Michiel van Baak

On 17:46, Thu 24 May 07, Khaled Chehab wrote:
> Any one knows where to install chan_sccp for asterisk 1.4 ???.
> 
> Please guide me from where can I download the asterisk 1.4 sccp channel
> driver and how to install it because I tried to get
> chan_sccp-mayday05.tar.gz
> 
> When I trying to install it ,error happened like this.
> 
> Please help me how  to solve this issue.

try chan_skinny that is in asterisk by default

-- 

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[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

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[asterisk-users] vmoutcall

2007-05-24 Thread Paul Aviles

Hello guys,

I have been looking for a way to call a cell phone after someone has left a
voice mail and allow the user to enter the mailbox password to listen to it
and found a very old entry of vmoutcall. I tried unsuccessfully to get it to
work with 1.4 and is beyond me. Has anyone gotten this to work or has a way
to accomplishing this? Most analog pbx's have this feature and I am amazed
Asterisk does not natively.

Any thoughts on getting similar functionality into the main code too?

Regards,

Paul Aviles


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[asterisk-users] Bottom line on fax reception

2007-05-24 Thread shadowym


 
So what is the bottom line?  Does it work or not.  I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.

What's the bottom line with recent updates on 1.2.x?  Is it production ready
for fax?  By production ready I mean that it just works all the time and
doesn't need any babysitting.  Do I have to worry about dropped lines,
sometimes not detecting incoming fax toneyada yada.  

I know I don't have to use fax on Asterisk but I really want to for various
reasons.  Mostly incoming but outgoing is a nice to have.  Should I use an
addon package and if so which one?  Any help would be appreciated. 

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Re: [asterisk-users] Integrated T1

2007-05-24 Thread Alex Balashov


On Thu, 24 May 2007, William Moore wrote:


On 5/24/07, Jeremy Mann <[EMAIL PROTECTED]> wrote:

Can an asterisk box equipped with a Digium T1 card handle Integrated T1
circuits?  I have a T1 with 768k data and the remaining channels voice, can
the asterisk box do the Data routing + Voice processing?


Yes, zaptel will create a device node for you.  Take a look at the
set-hdlc tool in zaptel and the less common channel types in the
default zaptel config file (rawhdlc is one, there are also others).


  Interesting!  Is this a relatively recent development?

  I stand corrected.

  Is there a way to make PPP-encapsulated T1 work as well?

--
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[asterisk-users] Re: Additional commands for MeetMeAdmin

2007-05-24 Thread Tony Mountifield

In article <[EMAIL PROTECTED]>,
Henry Cobb <[EMAIL PROTECTED]> wrote:
> Would anybody mind if the the following command options where added to
> MeetMeAdmin?

I'm sure no-one minds what you do to your own copies of Asterisk! :-)

> 0 - 9, * and #

Rather than create that many commands, why not have a d or D command
(for digit, or dtmf), and then the 0-9, *, #, A-D as a parameter to
that command?

> I'm considering hacking the code to add these commands to play the
> DTMFs to the specified user as tones and hope that the SIP or IAX
> channels then work with these correctly.

Hmmm, what problem are you trying to solve?

In Asterisk 1.4 and Trunk, there is an option to MeetMe for passing DTMF
frames through the conference. Your command should not generate tones, but
should use conf_queue_dtmf() to queue the dtmf into the conference. That way,
the DTMF is delivered to each participant channel as an asterisk DTMF frame,
and the channel can then handle it in a way appropriate to the technology
(e.g. Zap channels would re-generate tones, VoIP channels would deliver
DTMF control packets, etc).

Cheers
Tony
-- 
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Re: [asterisk-users] Integrated T1

2007-05-24 Thread William Moore

Here's a link that will get you most of the way there:
http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration
If you have any issues with setup, I recommend you contact Digium's
support to help you since I'm sure they've had the most experience
with it.

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

On Thu, 24 May 2007, Jeremy Mann wrote:

> Can an asterisk box equipped with a Digium T1 card handle Integrated T1
> circuits?  I have a T1 with 768k data and the remaining channels voice,
> can the asterisk box do the Data routing + Voice processing?

   The Zaptel/Asterisk infrastructure can definitely break particular
timeslots out of the T1 for voice, but it is not my impression that
any existing WAN drivers for Linux support Digium cards or cohabitation
with Zapata and can give you a serial data interface on other channels.

This e-mail, facsimile, or letter and any files or attachments transmitted with 
it contains information that is confidential and privileged. This information 
is intended only for the use of the individual(s) and entity(ies) to whom it is 
addressed. If you are the intended recipient, further disclosures are 
prohibited without proper authorization. If you are not the intended recipient, 
any disclosure, copying, printing, or use of this information is strictly 
prohibited and possibly a violation of federal or state law and regulations. If 
you have received this information in error, please notify Texas Health 
Management Group immediately at 1-817-310-4999. Texas Health Management Group, 
its subsidiaries, and affiliates hereby claim all applicable privileges related 
to this information.

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[asterisk-users] Re: meetme sounds

2007-05-24 Thread Tony Mountifield

In article <[EMAIL PROTECTED]>,
Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
> I am playing around with dynamic meetme conferences, and wanted to have 
> one person constantly in the conference, with calls "popping in and out".
> 
> Is there an option / any way of playing enter / leave sounds to the 
> person who created the conference only, and not the people leaving / 
> joining ?

Not without custom modifications to the code of app_meetme itself.

Cheers
Tony
-- 
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
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Re: [asterisk-users] Integrated T1

2007-05-24 Thread Alex Balashov


On Thu, 24 May 2007, Jeremy Mann wrote:

Can an asterisk box equipped with a Digium T1 card handle Integrated T1 
circuits?  I have a T1 with 768k data and the remaining channels voice, 
can the asterisk box do the Data routing + Voice processing?


  The Zaptel/Asterisk infrastructure can definitely break particular
timeslots out of the T1 for voice, but it is not my impression that
any existing WAN drivers for Linux support Digium cards or cohabitation
with Zapata and can give you a serial data interface on other channels.


This e-mail, facsimile, or letter and any files or attachments transmitted with 
it contains information that is confidential and privileged. This information 
is intended only for the use of the individual(s) and entity(ies) to whom it is 
addressed. If you are the intended recipient, further disclosures are 
prohibited without proper authorization. If you are not the intended recipient, 
any disclosure, copying, printing, or use of this information is strictly 
prohibited and possibly a violation of federal or state law and regulations. If 
you have received this information in error, please notify Texas Health 
Management Group immediately at 1-817-310-4999. Texas Health Management Group, 
its subsidiaries, and affiliates hereby claim all applicable privileges related 
to this information.

--
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--
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Re: [asterisk-users] Additional commands for MeetMeAdmin

2007-05-24 Thread Matt Florell


meetme DTMF doesn't work very well in SIP or IAX even if set to
inband. I had to resort to playing DTMF tones as audio files to get it
working for SIP and IAX, not a pretty thing.

MATT---

On 5/24/07, Henry Cobb <[EMAIL PROTECTED]> wrote:

Would anybody mind if the the following command options where added to
MeetMeAdmin?

0 - 9, * and #

I'm considering hacking the code to add these commands to play the
DTMFs to the specified user as tones and hope that the SIP or IAX
channels then work with these correctly.

-HJC
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Re: [asterisk-users] SCCP

2007-05-24 Thread Patrick

On Thu, 2007-05-24 at 17:46 -0700, Khaled Chehab wrote:
> Any one knows where to install chan_sccp for asterisk 1.4 ???.

Have you tried the chan_skinny driver that comes with 1.4?
Alternatively I saw a chan_sccp version for 1.4.3 or 1.4.4 here:
http://ting.ip-phone-forum.de/downloads.php?do=file&id=342

Regards,
Patrick

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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread 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

2007-05-24 Thread 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] 
> 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] - SOLVED - stream file not working but get data and exec background work

2007-05-24 Thread Eric \"ManxPower\" Wieling


Patrick Fortin wrote:

Hi

While testing I found a solution to my problem. I don't understand it 
maybe someone here can explain it.


In my script,
if I call a Playback just before my stream file then everything works ok.

Without the playback then the digits are not captured

I will playback a silence to patch my scripts.


Playback will issue an Answer before playing the audio.  Without the 
line being answered, you can't receive audio (or inband digits).

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RE: [asterisk-users] Asterisk Time Card

2007-05-24 Thread Shanon Swafford


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] 
> > 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

2007-05-24 Thread 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] 
> 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] meetme sounds

2007-05-24 Thread Steve Edwards


On Thu, 24 May 2007, Julian Lyndon-Smith wrote:

I am playing around with dynamic meetme conferences, and wanted to have one 
person constantly in the conference, with calls "popping in and out".


Is there an option / any way of playing enter / leave sounds to the person 
who created the conference only, and not the people leaving / joining ?


Not out of the box.

I did this for a customer a couple of years ago.

As I remember, the "key" was to add code to conf_run() to take the user 
out of the conference, play the custom sound file, and put them back into 
the conference. These in/out steps are needed to keep that user in sync 
with the conference. Otherwise, their audio will be offset by the length 
of the sound file.


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread 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] meetme sounds

2007-05-24 Thread Atlanticnynex


You can specify different options to start meetme with (announcements, etc.)
in the dialplan by having a separate extension for the person who wants to
here the sounds. I've never tried this, but I think it should work.

-kn0x

On 5/24/07, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:


I am playing around with dynamic meetme conferences, and wanted to have
one person constantly in the conference, with calls "popping in and out".

Is there an option / any way of playing enter / leave sounds to the
person who created the conference only, and not the people leaving /
joining ?

TIA

Julian.
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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread David Gomillion


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

2007-05-24 Thread Bruce Reeves

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]> 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]
> > 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

2007-05-24 Thread ram

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

On 5/24/07, Nitesh Divecha <[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.

with the AGI

you can do all

ram

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Re: [asterisk-users] Integrated T1

2007-05-24 Thread William Moore


On 5/24/07, Jeremy Mann <[EMAIL PROTECTED]> wrote:

Can an asterisk box equipped with a Digium T1 card handle Integrated T1
circuits?  I have a T1 with 768k data and the remaining channels voice, can
the asterisk box do the Data routing + Voice processing?


Yes, zaptel will create a device node for you.  Take a look at the
set-hdlc tool in zaptel and the less common channel types in the
default zaptel config file (rawhdlc is one, there are also others).
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RE: [asterisk-users] WiFi SIP phones

2007-05-24 Thread Shanon Swafford



I took the SIM card out of my Cingular Razor, installed it in the demo we
have, and it went right to work on GSM here in Dallas.

Here are some particulars I pulled from the web page below:

-Form Factor: Candy Bar Type (106Lx46Wx18H mm)
-LCD Display: 1.5", 128*128 pixel, 65K colors
-Protocols/Bands: GSM 900/1800/1900MHz
-Wi-Fi: 802.11b/g. 2.4 GHz
-Network Selection Choices: Wi-Fi Only / GSM Only / Wi-Fi Preferred / GSM
Preferred
-WLAN Security: WEP, WPA-PSK TKIP, WPA2-PSK AES
-VoIP Codecs: G.711, G.726, G.729
-Camera: 300k pixel VGA 
 
http://www.abptech.com/products/Pirelli/DPL10.html

Not sure about Linux, but our sales dept could find that out.

Regards,
Shanon
972-831-1600

-Original Message-
From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] 
Sent: Thursday, May 24, 2007 9:06 AM
To: Shanon Swafford
Cc: Asterisk-Users
Subject: RE: [asterisk-users] WiFi SIP phones


Has anyone installed Linux on your ABP phones, and got all
functionality (including GSM and WiFi)? Will these phones work in the US
(which radio frequency modes)?


On Thu, 2007-05-24 at 00:49 -0700,
[EMAIL PROTECTED] wrote:
> Date: Thu, 24 May 2007 00:10:23 -0500
> From: "Shanon Swafford" <[EMAIL PROTECTED]>
> Subject: RE: [asterisk-users] WiFi SIP phones
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="us-ascii"
> 
>  
> I work for ABP Technology and lurk on this list so I hope I'm not
> breaking
> any taboos...
>  
> ABP is now carrying a dual GSM/Wifi phone.  We tested 2 models, 1 had
> Windows-CE on it.  Some reason we only have the Non-CE version public
> right
> now.
>  
> http://www.abptech.com/products/Pirelli/DPL10.html
> http://www.abptech.com/products/Pirelli/DPL10.html>
> http://var.abptech.com/s.nl/it.A/id.2041/.f?sc=2&category=31> 
>  
> VARs/Resellers/ITSPs/Consultants:
> http://www.abptech.com/support/qa/index.php?target=become_reseller
>
http://www.abptech.com/support/qa/index.php?target=become_reseller
> > 
>  
> End Users go here and we'll help you find a place to buy one:
> http://www.abptech.com/aboutus/find_reseller.php
>  
> Shanon
> ABP Technology
> 
-- 

(C) Matthew Rubenstein



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[asterisk-users] Fwd: TC400B load problem

2007-05-24 Thread Arun Kumar


-- Forwarded message --
From: Arun Kumar <[EMAIL PROTECTED]>
Date: May 13, 2007 5:40 PM
Subject: TC400B load problem
To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users@lists.digium.com>

Hi

Im trying to install my TC400B trans coder card  when  I do:

modprobe wctc4xxp

tail -f /var/log/messages  says:

May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with
92 transcoders (srcs=000c, dsts=0101)
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with
92 transcoders (srcs=0101, dsts=000c)
May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps)
Transcoder support LOADED (firm ver = 56)
May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed with
error -5


please help

thanks

arun
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[asterisk-users] SCCP

2007-05-24 Thread Khaled Chehab

Any one knows where to install chan_sccp for asterisk 1.4 ???.

Please guide me from where can I download the asterisk 1.4 sccp channel
driver and how to install it because I tried to get
chan_sccp-mayday05.tar.gz

When I trying to install it ,error happened like this.

Please help me how  to solve this issue.

 

 

[EMAIL PROTECTED] t]# cd chan_sccp

[EMAIL PROTECTED] chan_sccp]# make clean

rm -rf chan_sccp.so .tmp

[EMAIL PROTECTED] chan_sccp]# make install

\Now compiling  chan_sccp.c 742 lines 

chan_sccp.c: In function `sccp_devicestate':

chan_sccp.c:133: error: `AST_DEVICE_UNKNOWN' undeclared (first use in this
function)

chan_sccp.c:133: error: (Each undeclared identifier is reported only once

chan_sccp.c:133: error: for each function it appears in.)

chan_sccp.c: In function `reload_config':

chan_sccp.c:397: warning: implicit declaration of function `ast_load'

chan_sccp.c:397: warning: assignment makes pointer from integer without a
cast

chan_sccp.c:555: error: incompatible type for argument 1 of `ast_inet_ntoa'

chan_sccp.c:555: error: too many arguments to function `ast_inet_ntoa'

chan_sccp.c:562: error: incompatible type for argument 1 of `ast_inet_ntoa'

chan_sccp.c:562: error: too many arguments to function `ast_inet_ntoa'

chan_sccp.c:566: error: incompatible type for argument 1 of `ast_inet_ntoa'

chan_sccp.c:566: error: too many arguments to function `ast_inet_ntoa'

chan_sccp.c:574: error: incompatible type for argument 1 of `ast_inet_ntoa'

chan_sccp.c:574: error: too many arguments to function `ast_inet_ntoa'

chan_sccp.c: In function `setcalledparty_exec':

chan_sccp.c:601: error: structure has no member named `type'

chan_sccp.c: At top level:

chan_sccp.c:666: warning: function declaration isn't a prototype

chan_sccp.c:700: warning: no previous prototype for 'reload'

chan_sccp.c:706: warning: function declaration isn't a prototype

chan_sccp.c:730: warning: function declaration isn't a prototype

chan_sccp.c:738: warning: function declaration isn't a prototype

chan_sccp.c:740: warning: function declaration isn't a prototype

make: *** [.tmp/chan_sccp.o] Error 1

 

 




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[asterisk-users] Integrated T1

2007-05-24 Thread Jeremy Mann

Can an asterisk box equipped with a Digium T1 card handle Integrated T1 
circuits?  I have a T1 with 768k data and the remaining channels voice, can the 
asterisk box do the Data routing + Voice processing?

It's only going to support 4-5 users(the voice channels won't all be active 
obviously).


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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread Mike Clark

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

2007-05-24 Thread David Gomillion


On 5/24/07, Alex Balashov <[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

2007-05-24 Thread Jon Pounder


Quoting Alex Balashov <[EMAIL PROTECTED]>:


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?



I think he means the prepaid phone card discussions



--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
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Jon Pounder

   _/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
_/_/_/  _/  _/ _/_/_/  _/  _/_/
   _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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[asterisk-users] Additional commands for MeetMeAdmin

2007-05-24 Thread Henry Cobb


Would anybody mind if the the following command options where added to
MeetMeAdmin?

0 - 9, * and #

I'm considering hacking the code to add these commands to play the
DTMFs to the specified user as tones and hope that the SIP or IAX
channels then work with these correctly.

-HJC
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[asterisk-users] meetme sounds

2007-05-24 Thread Julian Lyndon-Smith

I am playing around with dynamic meetme conferences, and wanted to have 
one person constantly in the conference, with calls "popping in and out".


Is there an option / any way of playing enter / leave sounds to the 
person who created the conference only, and not the people leaving / 
joining ?


TIA

Julian.
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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread David Gomillion


On 5/24/07, Nitesh Divecha <[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] PSP Voip

2007-05-24 Thread Matthew Rubenstein

What was the content of the message you sent?

And what is the deal with these messages the list delivers "scrubbed" of
their content? Maybe the listbot can't handle "multipart/alternative"
MIME messages.


On Thu, 2007-05-24 at 05:52 -0700,
[EMAIL PROTECTED] wrote:
> Date: Thu, 24 May 2007 08:50:58 -0400
> From: "Dean Collins" <[EMAIL PROTECTED]>
> Subject: [asterisk-users] PSP Voip
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="us-ascii"
> 
> Skipped content of type multipart/alternative-- next part
> --
> A non-text attachment was scrubbed...
> Name: not available
> Type: image/gif
> Size: 2775 bytes
> Desc: image001.gif
> Url :
> http://lists.digium.com/pipermail/asterisk-users/attachments/20070524/7908e840/attachment.gif
>  
-- 

(C) Matthew Rubenstein

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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread Alex Balashov



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.  :-)

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
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[asterisk-users] SLA with SIP-only environment

2007-05-24 Thread Yehavi Bourvine +972-8-9489444

Hello,

  All the examples of SLA talk about Zap channels from one side and SIP on the
other side, while my system is a pure SIP one.

I would like to have two phones having extensions 1 & 2 defined on them, and
when someone calls extension 1 it rings on both, each one can see its status,
and when one station puts line 1 on hold the other one can pick it.  Is it
possible at all?  If so, can someone give the relevant fragments for sla.conf,
sip.conf and extensions.conf?


 Thanks! __Yehavi:
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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread Alex Balashov


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

2007-05-24 Thread 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] 
> 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] WiFi SIP phones

2007-05-24 Thread Matthew Rubenstein

Has anyone installed Linux on your ABP phones, and got all
functionality (including GSM and WiFi)? Will these phones work in the US
(which radio frequency modes)?


On Thu, 2007-05-24 at 00:49 -0700,
[EMAIL PROTECTED] wrote:
> Date: Thu, 24 May 2007 00:10:23 -0500
> From: "Shanon Swafford" <[EMAIL PROTECTED]>
> Subject: RE: [asterisk-users] WiFi SIP phones
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="us-ascii"
> 
>  
> I work for ABP Technology and lurk on this list so I hope I'm not
> breaking
> any taboos...
>  
> ABP is now carrying a dual GSM/Wifi phone.  We tested 2 models, 1 had
> Windows-CE on it.  Some reason we only have the Non-CE version public
> right
> now.
>  
> http://www.abptech.com/products/Pirelli/DPL10.html
> http://www.abptech.com/products/Pirelli/DPL10.html>
> http://var.abptech.com/s.nl/it.A/id.2041/.f?sc=2&category=31> 
>  
> VARs/Resellers/ITSPs/Consultants:
> http://www.abptech.com/support/qa/index.php?target=become_reseller
> http://www.abptech.com/support/qa/index.php?target=become_reseller
> > 
>  
> End Users go here and we'll help you find a place to buy one:
> http://www.abptech.com/aboutus/find_reseller.php
>  
> Shanon
> ABP Technology
> 
-- 

(C) Matthew Rubenstein

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Re: [asterisk-users] WiFi SIP phones

2007-05-24 Thread Michael Graves

So many hotspots require some form of web based login, possibly even with 
acceptance of a browser cookie. That takes a lot fo wifi SIP devices out of the 
game. The best sucess I had 
while travelling with the WIP5000 in my bag was also carrying an Apple Airport 
Express. This allowed me to use wired network access in hotels on a wlan with 
known properties.

However, while it was an interesting project, it was never as practical as my 
cell phone.

Michael

On Thu, 24 May 2007 08:00:54 -0400, SIP wrote:

>I've gotten SIP calls to work via hotspots on a Dell Axim running 
>SJ-Phone. I've also had reasonable success with a Nokia E60. I've had 
>ZERO luck from a hotspot on the UTStarCom phones, nor on the Linksys 
>wifi phones. I'm not quite sure yet why something which is ONLY a wifi 
>phone has more issues connecting to wifi hotspots than, say, 
>multi-function devices.

>However, looking at the shortcuts both developers have taken in their 
>firmware, I'm of the opinion it's just sloppy code.


>Michael Graves wrote:
>> I travel a lot for work. I frequently find hotels that have wifi, free 
>> or otherwise available. But I've yet to find it anywhere near 
>> sufficient to support voip applications. At least not good enough to 
>> compel me to not use my cell phone. If you have control of the host 
>> LAN then you can ensure it meets the needs of a wifi SIP phone, 
>> otherwise why bother.
>>
>> Has anyone ever seen anyone making a voip call on a wif handset ata 
>> public hotspot? While that would score many geek points I doubt it 
>> would work in many places.
>>
>> About 18 mo ago I bought the Hitachi Cable WIP5000 handset. It was 
>> seriously flawed so I resold it after a few months and settled on the 
>> Aastra desk phone. I do wish the cordless handsets were a little more 
>> like a Panasonic cordless phone...more buttons...easier to program, etc.
>>
>> Michael
>>
>> On Wed, 23 May 2007 21:59:03 -0400, Justin Moore wrote:
>>
>> >On 5/23/07, Michael Graves <[EMAIL PROTECTED]> wrote:
>> >> I must say that I've VERY happy with my Aastra 4801 CT phones. I 
>> think that
>> >> they're DECT. Each can have up to six cordless handsets. 
>> Technically its a 9
>> >> line phone, but if you use G.729 you can only sustain two calls at 
>> once. I
>> >> can have a call on the portable and easily take another on the base.
>> >
>> >I am also an extremely happy user of an Aastra 480i CT. Awesome phone.
>> >However, I was under the impression that the OP was looking for a WiFi
>> >phone that could be carried from place to place, but I may be wrong...
>> >
>> >--
>> >Justin Moore
>> >aka wantmoore
>> >---
>> >_www.wantmoore.com_
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>> >
>>
>>
>> 
>>
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[asterisk-users] Nokia release

2007-05-24 Thread Rizwan Hisham


Hi all,
sorry to ask you something not related to asterisk, but i really want to
know whether the Nokia N95 cell phone is released in the USA or not? if
somebody from USA knows, plz reply.

--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] PRI problem, pri_fixup_principle: Call specified, but not found?

2007-05-24 Thread Carlos G Mendioroz

Here...
Please advise if any special flags/options are needed.

< [ 02 01 01 27 ]

< Supervisory frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 019 P/F: 1
< 0 bytes of data
-- ACKing all packets from 18 to (but not including) 19
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
-- Unsolicited RR with P/F bit, responding
Sending Receiver Ready (51)

> [ 02 01 01 67 ]

> Supervisory frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 051 P/F: 1
> 0 bytes of data
-- Restarting T203 counter
-- Restarting T203 counter
T203 counter expired, sending RR and scheduling T203 again
Sending Receiver Ready (51)

> [ 00 01 01 67 ]

> Supervisory frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 051 P/F: 1
> 0 bytes of data
-- Restarting T203 counter

< [ 00 01 01 27 ]

< Supervisory frame:
< SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 019 P/F: 1
< 0 bytes of data
-- ACKing all packets from 18 to (but not including) 19
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
-- Got RR response to our frame
-- Restarting T203 counter

< [ 02 01 01 b1 ]

< Supervisory frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 088 P/F: 1
< 0 bytes of data
-- ACKing all packets from 87 to (but not including) 88
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
-- Unsolicited RR with P/F bit, responding
Sending Receiver Ready (6)

> [ 02 01 01 0d ]

> Supervisory frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 006 P/F: 1
> 0 bytes of data
-- Restarting T203 counter
-- Restarting T203 counter
T203 counter expired, sending RR and scheduling T203 again
Sending Receiver Ready (6)

> [ 00 01 01 0d ]

> Supervisory frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 006 P/F: 1
> 0 bytes of data
-- Restarting T203 counter

< [ 00 01 01 b1 ]

< Supervisory frame:
< SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 088 P/F: 1
< 0 bytes of data
-- ACKing all packets from 87 to (but not including) 88
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
-- Got RR response to our frame
-- Restarting T203 counter
-- Accepting AUTHENTICATED call from 10.8.0.6:
   > requested format = alaw,
   > requested prefs = (),
   > actual format = alaw,
   > host prefs = (),
   > priority = mine
-- Executing Dial("IAX2/10.8.0.6:4569-3", "Zap/g2/113") in new stack
-- Requested transfer capability: 0x00 - SPEECH

> [ 00 01 26 66 08 02 0e 89 05 04 03 80 90 a3 18 04 e9 82 83 81 28 08 43
61 72 6c 6f 73 20 4d 6c 06 41 81 31 31 30 30 70 04 c1 31 31 33 a1 ]

> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> N(S): 019   0: 0
> N(R): 051   P: 0
> 41 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8)  len=41
> Call Ref: len= 2 (reference 3721/0xE89) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
>  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
>  Ext: 1  User information layer 1: A-Law (35)
> [18 04 e9 82 83 81]
> Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, Exclusive
Dchan:0
>ChanSel: Reserved
>   Ext: 1  DS1 Identifier: 2
>   Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
>   Ext: 1  Channel: 1 ]
> [28 08 43 61 72 6c 6f 73 20 4d]
> Display (len= 8) [ Carlos M ]
> [6c 06 41 81 31 31 30 30]
> Calling Number (len= 8) [ Ext: 0  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>   Presentation: Presentation permitted, user
number passed network screening (1) '1100' ]
> [70 04 c1 31 31 33]
> Called Number (len= 6) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '113' ]
> [a1]
> Sending Complete (len= 1)
-- Called g2/113

< [ 00 01 01 28 ]

< Supervisory frame:
< SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 020 P/F: 0
< 0 bytes of data
-- ACKing all packets from 18 to (but not including) 20
-- ACKi

Re: [asterisk-users] Asterisk Clusters

2007-05-24 Thread ram


On 5/23/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:


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?




Hi

why dont you looking
this kind of solution

DS3TDMOVERIPSER-ASterisk

ram

Thanks,

Nitesh


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Re: [asterisk-users] Asterisk Time Card

2007-05-24 Thread ram


On 5/24/07, Nitesh Divecha <[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] Parking Lot CallerID

2007-05-24 Thread Alex Balashov


On Thu, 24 May 2007, Ken Williams wrote:


Beings the call is originating on the phone I'm not sure there's a way
to push the CID back to it, any thoughts?


  It is possible that the phone accepts some form of SIP NOTIFY message 
for revising its caller ID display.  Some caller ID / CNAM is implemented 
this way because the lookup takes place subsequent to the establishment of 
end-to-end signaling parameters, mirroring the ISDN setting for the same.


--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
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[asterisk-users] Parking Lot CallerID

2007-05-24 Thread Ken Williams

Is there anyway of storing an incoming calls CallerID on a parked call
and having it restored when someone picks up the parked call?
 
I've tried storing the CID as a global variable and restoring it in my
dialplan, and while NoOp shows it working, the phone ignores it and uses
the parking lot extension for callerid instead.  I believe this is
because the phone is calling out instead of a call coming in, is there
anyway around this?
 
This is a basic idea of what I've done to try to capture the CID in
testing:
 
exten => 200,1,SetGlobalVar(P1NAME=${CALLERID(NAME)})
exten => 200,n,SetGlobalVar(P1NUM=${CALLERID(NUM)})
exten => 200,n,Park()
exten => _20x,1,Wait(1)
exten => _20x,n,NoOp(${DIAL_OPTIONS})
exten => _20x,n,Set(CALLERID(NAME)=${P1NAME})
exten => _20x,n,Set(CALLERID(NUM)=${P1NUM})
exten => _20x,n,NoOp(${CALLERID(NAME)} ${CALLERID(NUM)})
exten => _20x,n,ParkedCall(${EXTEN})

Beings the call is originating on the phone I'm not sure there's a way
to push the CID back to it, any thoughts?
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Re: [asterisk-users] Re: Kernel Panic in wct4xxp during unload on Zaptel-1.4.4

2007-05-24 Thread James FitzGibbon


On 5/22/07, Axel Thimm <[EMAIL PROTECTED]> wrote:


Have you tried using the 1.4.x atrpms packages?




I  did try the 1.4 packages from atrpms overnight yesterday, with similar
results.  I was able to address the kernel panic when unloading by
commenting out "ztcfg -s" in the stop() function of the init script (based
on suggestions on this list).

The system appeared stable (went through several clean startup/shutdown
cycles), but then proceeded to kernel panic four times in three hours when
calls were being processed, forcing me to downgrade to 1.2 again.

Unfortunately, I was remote and unable to capture the full kernel panic
details, so I'm kind of stuck at square one until I can upgrade again during
a maint window and attempt to force a panic during call processing.

--
j.
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[asterisk-users] bridging calls between two numbers with extensions

2007-05-24 Thread Matthew M. Boedicker

I am using the Originate Manager API call to bridge a call between two
numbers. The first number is called by originate and the action is set
to Dial the other number.

If the second number has an extension that needs to be sent using DTMF
I can use the D option when I call Dial like this:

Local/[EMAIL PROTECTED]||D(ww1ww2ww3ww4)

This works fine. My question is how do I send DTMF to the first number
called? As far as I know there is no way to send DMTF using Originate like
you can with Dial.

Thanks,
Matthew Boedicker
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[asterisk-users] PSP Voip

2007-05-24 Thread Dean Collins

This isn't a serious post but

 

Interesting announcement yesterday about PSP's being hooked up with voip
services to not only their 'home' pc's but also public wifi hotspots.

http://gigaom.com/2007/05/23/sony-psp-voip-usa/

 

Anyone have any further information? Are they allowing inbound call
services as well or outbound only? 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

    
 
 

 

 

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[asterisk-users] Asterisk Time Card

2007-05-24 Thread 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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Re: [asterisk-users] CDR on channel 'IAX2/u92613106-3' already started

2007-05-24 Thread Steve Murphy



On Wed, 2007-05-23 at 20:51 -0600, Mike Diehl wrote:
> Hi all,
> 
> I'm having a problem with an asterisk server being unable to call certain 
> cellphones and answering machines.  Anytime the person answers the phone 
> call, everything works well.  But when the call goes to voicemail or an 
> answering machine, I get the error message below:
> --- snip ---

Mike--

I assume you're using 1.4 or trunk; normally this shouldn't be a big
deal. Exactly which version of 1.4 are you using? Have you tried the
latest SVN version? I made some fixes concerning check_start.

murf


-- 
Steve Murphy <[EMAIL PROTECTED]>
Digium

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RE: [asterisk-users] PRI problem, pri_fixup_principle: Call specified, but not found?

2007-05-24 Thread John Treble

> in a PRI setup, the receiving side is changing the B channel at
proceeding.

Please post the layer 3 trace for this call scenario so that members of this
forum can help you debug your problem.


John Treble
Ottawa, Canada


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Carlos G Mendioroz
> Sent: May 24, 2007 7:53 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] PRI problem, pri_fixup_principle: Call
> specified,but not found?
> 
> Hi,
> in a PRI setup, the receiving side is changing the B channel at
> proceeding. It seems this sometimes breaks some logic
> (pri_fixup_principle) and then the hangup kind of breaks, release is not
> answered and a restart cycle is triggered (by remote side).
> 
> Anyone can help me debug this ? I've seen many posts with simmilar
> issues but no answer/solution.
> 
> This is happening on Asterisk 1.2.16 + libpri 1.2.4 on a sangoma A104D.
> On a general side, where can I find a document (other than sources :)
> to start digging where the Zap->pri mapping is done and how,
> or at leaset what pri_fixup_principle is supposed to do...
> 
> TIA,
> --
> Carlos G Mendioroz  <[EMAIL PROTECTED]>
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Re: [asterisk-users] WiFi SIP phones

2007-05-24 Thread SIP

I've gotten SIP calls to work via hotspots on a Dell Axim running 
SJ-Phone. I've also had reasonable success with a Nokia E60. I've had 
ZERO luck from a hotspot on the UTStarCom phones, nor on the Linksys 
wifi phones. I'm not quite sure yet why something which is ONLY a wifi 
phone has more issues connecting to wifi hotspots than, say, 
multi-function devices.

However, looking at the shortcuts both developers have taken in their 
firmware, I'm of the opinion it's just sloppy code.

Michael Graves wrote:
I travel a lot for work. I frequently find hotels that have wifi, free 
or otherwise available. But I've yet to find it anywhere near 
sufficient to support voip applications. At least not good enough to 
compel me to not use my cell phone. If you have control of the host 
LAN then you can ensure it meets the needs of a wifi SIP phone, 
otherwise why bother.

Has anyone ever seen anyone making a voip call on a wif handset ata 
public hotspot? While that would score many geek points I doubt it 
would work in many places.

About 18 mo ago I bought the Hitachi Cable WIP5000 handset. It was 
seriously flawed so I resold it after a few months and settled on the 
Aastra desk phone. I do wish the cordless handsets were a little more 
like a Panasonic cordless phone...more buttons...easier to program, etc.

Michael

On Wed, 23 May 2007 21:59:03 -0400, Justin Moore wrote:

>On 5/23/07, Michael Graves <[EMAIL PROTECTED]> wrote:
>> I must say that I've VERY happy with my Aastra 4801 CT phones. I 
think that
>> they're DECT. Each can have up to six cordless handsets. 
Technically its a 9
>> line phone, but if you use G.729 you can only sustain two calls at 
once. I

>> can have a call on the portable and easily take another on the base.
>
>I am also an extremely happy user of an Aastra 480i CT. Awesome phone.
>However, I was under the impression that the OP was looking for a WiFi
>phone that could be carried from place to place, but I may be wrong...
>
>--
>Justin Moore
>aka wantmoore
>---
>_www.wantmoore.com_
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Re: [asterisk-users] modprobe

2007-05-24 Thread Josu Lazkano


Thanks Giorgio!!!

I made "modprobe zaptel" and then "ztcfg -vv" anI have this:

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: D-channel (Default) (Slaves: 03)

3 channels configured.

ZT_SPANCONFIG failed on span 1: No such device or address (6)

I think is better but not enough, thanks for that.

Anyone uses the Billion ISDN PCI?

Thanks every body!!!


2007/5/24, Giorgio Incantalupo <[EMAIL PROTECTED]>:


Hi Josu,
I had the same problem with wctdm.I just loaded zaptel before wctdm
and it was all ok.
Hope it can help you.  :)

Giorgio Incantalupo


Josu Lazkano wrote:
> Hello every boy again
>
> I have some problems with modprobe. When I type "modprobe zaphfc",
> this error happens "FATAL: Module zaphfc not found."
>
> And when I tyoe "ztcfg -vv" this error happens:
>
> Notice: Configuration file is /etc/zaptel.conf
> line 0: Unable to open master device '/dev/zap/ctl'
>
> 1 error(s) detected
>
> Someone can help me???
>
> Thanks to all.
> 
>
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--

_
Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
FG&A srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172

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[asterisk-users] PRI problem, pri_fixup_principle: Call specified, but not found?

2007-05-24 Thread Carlos G Mendioroz

Hi,
in a PRI setup, the receiving side is changing the B channel at
proceeding. It seems this sometimes breaks some logic
(pri_fixup_principle) and then the hangup kind of breaks, release is not
answered and a restart cycle is triggered (by remote side).

Anyone can help me debug this ? I've seen many posts with simmilar
issues but no answer/solution.

This is happening on Asterisk 1.2.16 + libpri 1.2.4 on a sangoma A104D.
On a general side, where can I find a document (other than sources :)
to start digging where the Zap->pri mapping is done and how,
or at leaset what pri_fixup_principle is supposed to do...

TIA,
-- 
Carlos G Mendioroz  <[EMAIL PROTECTED]>
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Re: [asterisk-users] modprobe

2007-05-24 Thread Tzafrir Cohen

On Thu, May 24, 2007 at 12:23:07PM +0200, Josu Lazkano wrote:
> Hello john, thanks for response.
> 
> I am trying to install a Billion ISDN on Asterisk
> 
> I have Debian Etch and I installed theese packages:
> 
> apt-get install linux-headers-`uname -r`
> apt-get install make
> 
> apt-get install ncurses-base ncurses-bin ncurses-term

[ snip]

Yeah. That sounds familiar...

You seem to have ignored my response to your previous post.

So a short summary:

A. it would be much faster for you to set up a system with standard 
   packages from Debian Etch

B. The package you're trying to use actually pulls lder zaptel/asterisk
   than the ones in Etch. See that previous post for links to newer bristuff.

C. You had a problem installing zaptel . Read the README there regarding 
   setting up the /usr/src/linux link that nobody really needs except 
   the bristuff build script.

Ignoring previous replies and starting totally new threads is *not* a 
way to gin popularity here. I also hope you realise that it is a good 
method to get a bunch of irrelevant replies.

Please follow-up on the original thread. Reply to your or my message 
there so mail programs will consider it part of the same thread.

-- 
   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] modprobe

2007-05-24 Thread Tzafrir Cohen

On Thu, May 24, 2007 at 12:16:26PM +0200, Giorgio Incantalupo wrote:
> Hi Josu,
> I had the same problem with wctdm.I just loaded zaptel before wctdm 
> and it was all ok.
> Hope it can help you.  :)

Actually, you don't need to modprobe zaptel explicitly. You just need 
to not run ztcfg at modprobe time.

On some systems (notably RHEL4/centos4, and probably some matching 
fedoras) generation of the udev nodes was horribly slow, and hence 
you need to wait a few seconds after you insmod zaptel (which happens
implicitly when you modproe any other zaptel module) till you can run 
ztcfg.

But this is not really his problem. In his case zaptel did not get loaded 
at all.

-- 
   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] modprobe

2007-05-24 Thread Josu Lazkano


Hello Tzafrir, thanks for response.

I am trying to install a Billion ISDN on Asterisk

I have Debian Etch and I installed theese packages:

apt-get install linux-headers-`uname -r`
apt-get install make


apt-get install ncurses-base ncurses-bin ncurses-term

apt-get install libncurses5 libncurses5-dev
apt-get install bison
apt-get install openssl
apt-get install libssl0.9.8
apt-get install libssl-dev


apt-get install libeditline0 libeditline-dev libedit-dev libedit2

apt-get install gcc
apt-get install zlib1g-dev


To install Asterisk with Bristuff I do that:

in usr/src:

wget 
http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-current.tar.gz

tar zxvf bristuff-0.3.0-current.tar.gz
cd bristuff-0.3.0-PRE-1r
./install.sh

That could help?

Thanksss

2007/5/24, Tzafrir Cohen <[EMAIL PROTECTED]>:


On Thu, May 24, 2007 at 11:17:57AM +0200, Josu Lazkano wrote:
> Hello every boy again
>
> I have some problems with modprobe. When I type "modprobe zaphfc", this
> error happens "FATAL: Module zaphfc not found."

zaphfc is part of bristuff. have you installed brisuff (or any other
bristuffed zaptel package, such as the one from Debian)?

>
> And when I tyoe "ztcfg -vv" this error happens:
>
> Notice: Configuration file is /etc/zaptel.conf
> line 0: Unable to open master device '/dev/zap/ctl'

This is normally an indication that the module zaptel is not loaded.
Which makes sense, as the driver you modprobed for did not exist nd hence
could not pull zaptel with it.

What zaptel hardware do you have? How have you installed Zaptel?

--
   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] modprobe

2007-05-24 Thread Josu Lazkano


Hello john, thanks for response.

I am trying to install a Billion ISDN on Asterisk

I have Debian Etch and I installed theese packages:

apt-get install linux-headers-`uname -r`
apt-get install make

apt-get install ncurses-base ncurses-bin ncurses-term

apt-get install libncurses5 libncurses5-dev
apt-get install bison
apt-get install openssl
apt-get install libssl0.9.8
apt-get install libssl-dev

apt-get install libeditline0 libeditline-dev libedit-dev libedit2

apt-get install gcc
apt-get install zlib1g-dev


To install Asterisk with Bristuff I do that:

in usr/src:

wget 
http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-current.tar.gz

tar zxvf bristuff-0.3.0-current.tar.gz
cd bristuff-0.3.0-PRE-1r
./install.sh

That could help?

Thanksss

2007/5/24, John covici <[EMAIL PROTECTED]>:


We would need more details to help -- version of asterisk and zaptel
and what you did to try to install them -- hardware you have, etc and
why you did that modprobe statement.


on Thursday 05/24/2007 Josu Lazkano([EMAIL PROTECTED]) wrote
> Hello every boy again
>
> I have some problems with modprobe. When I type "modprobe zaphfc", this
> error happens "FATAL: Module zaphfc not found."
>
> And when I tyoe "ztcfg -vv" this error happens:
>
> Notice: Configuration file is /etc/zaptel.conf
> line 0: Unable to open master device '/dev/zap/ctl'
>
> 1 error(s) detected
>
> Someone can help me???
>
> Thanks to all.
> Hello every boy againI have some problems with modprobe. When I
type "modprobe zaphfc", this error happens "FATAL: Module
zaphfc not found."And when I tyoe "ztcfg -vv" this
error happens:
> Notice: Configuration file is /etc/zaptel.confline 0: Unable
to open master device '/dev/zap/ctl'1 error(s)
detectedSomeone can help me???Thanks to all.
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How do
you spend it?

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 [EMAIL PROTECTED]
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Re: [asterisk-users] modprobe

2007-05-24 Thread Giorgio Incantalupo


Hi Josu,
I had the same problem with wctdm.I just loaded zaptel before wctdm 
and it was all ok.

Hope it can help you.  :)

Giorgio Incantalupo


Josu Lazkano wrote:

Hello every boy again

I have some problems with modprobe. When I type "modprobe zaphfc", 
this error happens "FATAL: Module zaphfc not found."


And when I tyoe "ztcfg -vv" this error happens:

Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

Someone can help me???

Thanks to all.


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--

_
Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
FG&A srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172  


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