[Asterisk-Users] Dial Limit Call Options

[Alejandro G]

Hi,

Is there a way to know if after using the Dial command and specifying
L(X:Y:Z) option for limiting the duration of the call and if the calls
reachs that limit have an indication that the caller reachs the limit? (i.e.
DIALSTATUS)

Thanks


Alejandro Ghergherian

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RE: [Asterisk-Users] Voicemail as an email attachement

[Goran Skular]

Small.. just app_voicemail.c and a sendEmail script...

You can download it from here:

app_voicemail.c
http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f
ileinfoid=9
and


sendEmail
http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f
ileinfoid=10




sendEmail is most important.. code change is really small in app_voicemail..
but here it is..


1. install sendEmail

2. Edit app_voicemail.c :


You will need to change app_voicemail.c to suit your needs.. Go to line 1035
(or find goran.skular) and:

Change [EMAIL PROTECTED] to from address you want to show up

Mail.slsolucije.hr:25 change to your mail.server.xxx:smtp 

Password_here is place for your password..


Go to line 1130 also (or find next appereance of goran.skular) and to the
same again.


That's all in short.

Have a nice day.
 

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, October 19, 2005 4:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail as an email attachement

Yes. I am interested. I will make provisions for the upload. How big are
the files?

Thanks

BEN

Goran Skular wrote:
 I changed my app_voicemail.c to work not with sendmail but with sendEmail
 that connects to any SMTP and sends email with attachment...

 It's dirty, but it works.

 If you are interested I can upload app_voicemail.c and sendEmail package
 somewhere..



I have configured the voicemail.conf file as per the wiki to email
voicemails as an attachment. I cannot find any instructions/locations to
set the outgoing server login information. Furthermore, I can get no
emails from asterisk. Can anyone point me to the next step to setup the
attachment of voicemail messages to an email?

Thanks

BEN
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[Asterisk-Users] Re: SNOM 360 Unknown SIP command 'PUBLISH'

[Shanon Swafford]

Without seeing the actual SIP Message.  I'm guessing it is Number Guessing.  
It is on default on Snom phones.

Regards,
Shanon

[EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]...
 Hi List
 
  
 
 I'm getting this notification from my one and only SNOM 360 every time 
 a number button is pushed.
 
 I know that it's only a notification, but it really irritates me. Is 
 it anything I can/should do anything about ??
 
  
 
 Oct 12 10:34:33 NOTICE[3566]: chan_sip.c:10530 handle_request: Unknown 
 SIP command 'PUBLISH' from '192.168.100.100'
 
  
 
  
 
 By the way I'm using * 1.0.9 CVS-HEAD September 15. 2005
 
  
 
 Best regards
 
  
 
 BennyBad
 
 


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RE: [Asterisk-Users] Voicemail as an email attachement

[Goran Skular]

On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote:
 I have configured the voicemail.conf file as per the wiki to email
 voicemails as an attachment. I cannot find any instructions/locations to
 set the outgoing server login information. Furthermore, I can get no
 emails from asterisk. Can anyone point me to the next step to setup the
 attachment of voicemail messages to an email?

Set up a sendmail. Or basically: an MTA. Any linux distro comes with
at least one (postfix seems to be the preffered choice nowadays). Which
one do you use?

There are a bunch of programs that provide /usr/sbin/sendmail but don't
spool the result. Check msmtp, ssmtp, masqmail and nullmailer. There are
probably others.

The downside is that messages that have, for some reason, not been
delivered in the first shot (e.g: due to some transient network error)
will be dropped rather than queued.


I was playing with mta, but this is so complicated, specially if you are on
dynamic ip address, so it is much easier to use smtp for sending mails..

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RE: [Asterisk-Users] Connection question

[Tomislav Parcina]

As far as I know you can. The only thing you need to know is what ports does 
your Alcatel PBX use.


Tomislav
 

 Asterisk seems to be a very good peace of software, but i am 
 interested to know if i can use plain ISDN cards with it, i 
 mean use the isdn cards as a passthrough device between my 
 alcatel pbx and voip users.
 
 thanks
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Re: [Asterisk-Users] chan_capi-0.6 configuration Query with Eicon Diva 4BRI

[Voicomm User]

So, are you saying 'msn=' parameter is not required for both Point to
Point and Point to Multi Point?

thanks
-r

On 10/20/05, John Daragon [EMAIL PROTECTED] wrote:
 Voicomm User wrote:
  Hello
 
  Hardware: Eicon Diva 4BRI ISDN Card
  Software : Asterisk : Asterisk CVS-v1-0-08/13/05-19:51:52
 Chan Capi: chan_capi-0.6
 
  We are using an Eicon 4BRI ISDN Card here in Australia with Asterisk,
  connected to 4 OnRamp services with Telstra.
 
  There are 8 available channels, but after upgrading to latest capi
  driver we notice that the box is not able to handle more than 2 calls
  at the same time. An engaged signal is heard at the other end. After
  this happens once, some calls fail even when all channels are free.
  I don't see any messages on console for failes calls. Even when I turn
  on 'capi debug' and 'set verbose 20'.
 
  The telstra personnel have confirmed busy signal is sent out by the
  PABX. But its bizarre not to see any messages. No error messages are
  logged as well.
 
  capi info :
  Contr1: 2 B channels total, 2 B channels free.
  Contr2: 2 B channels total, 2 B channels free.
  Contr3: 2 B channels total, 2 B channels free.
  Contr4: 2 B channels total, 2 B channels free.
 
  capi.conf
 
  [general]
  nationalprefix=0
  internationalprefix=00
  rxgain=0.8
  txgain=0.8
 
  [isdn]
  isdnmode=ptp ; Is this correct for Point to Point Mode?
  msn=8 digit local number
  group=1
  incomingmsn=*
  controller=1,2,3,4 ; there are 4 controllers
  devices=2   ; should this be 8?
  softdtmf=on
  relaxdtmf=on
  accountcode=
  context=main-menu
  echocancelold=yes
  ;echocancel=yes  ; Turning this on gives a error message each time a
  call is terminated.
  usecallerid=yes
  callerid=asreceived
  ;echosquelch=1
  ;echotail=64
  ;callgroup=1
  ;pickupgroup=1

 The syntax has changed a bit. Time was when the devices= line
 basically said OK, that's this controller done with, let's commit that
 and start on the next one...  With 0.6 (if I read it correctly) it goes :

 [general]
 .
 .

 [some_string]

 group=1
 isdnmode=did   -- note this has changed  [DID/MSN]
 incomingmsn=*
 rxgain=1.0
 txgain=0.8
 controller=1
 softdtmf=0
 accountcode=
 context=from-pstn
 echosquelch=0
 echocancel=yes
 echotail=64
 devices=2

 [some_other_string]

 group=1
 isdnmode=did
 incomingmsn=*
 rxgain=1.0
 txgain=0.8
 controller=2
 softdtmf=0
 accountcode=
 context=from-pstn
 echosquelch=0
 echocancel=yes
 echotail=64
 devices=2

 Hope this helps...

 jd


 --

 John Daragon  [EMAIL PROTECTED]
 argv[0] limited
 Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
 v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127


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[Asterisk-Users] Asterisk in Croatia - Zagreb

[Goran Skular]

Hello,



I was wondering how many people from Croatia are
using and playing with Asterisk. Recently I had a contact with one user and I
am very glad.

It will be really nice to organize a Croatian
Asterisk community and on that way we are organizing a little gathering.

It does not matters how much experience you have,
everthing you need is some interest in Asterisk.

Beside my last contact I know that croatian wifi community
ZG Wireless is using Asterisk also.



So, 



Everyone of you, located in Croatia, please contact me here or
on email.



For the purpose of collecting as much people, gathering
is to be expected next month (around 19th)



Send me an e-mail or even register on www.migo-systems.com. Further info will
be available later.



Looking forward for it,



Goran Skular

www.slsolucije.hr










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[Asterisk-Users] Chan-capi sound choppy

[Mario Fernández Alonso]

Hi all.

I'm using Debian Sarge with Asterisk  1.0.7.dfsg.1-2 and Asterisk-chan-capi 
0.3.5-11 on a P-III 800 with 196MB RAM.
The isdn card is AVM B1 isa and the softphone is eyeBeam 1.1 3004t stamp 16741.
The audio codec G711aLaw works so fine for me. Other codecs sounds too bad.
The problem comes when I use the two B channels of isdn card. The sound is 
choppy, but if I use only one channel the audio is good.
The card is the only card using IRQ 5. The machine at the moment of sound 
choppy is 70% idle and 55MB RAM free.
I had download the source package of Asterisk-chan-capi, and changing  
AST_CAPI_MAX_B3_BLOCK_SIZE from 160 to 400 the problem of sound choppy is 
nearly solved.
But, that is the way?
Thanks.
---
Mario Fdez. Alonso
Abysal Systems
Parque Emp. Las Rozas
Jose Echegaray, 5
28230 Madrid
Tfl: 916404437
Fax: 916403119
[EMAIL PROTECTED]
www.abysal.com
---



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[Asterisk-Users] Can't build Asterisk on SuSE

[telephony]

SuSE Linux Enterprise Server 9
Asterisk 1.2.0 beta1

I am trying to build and install Asterisk on SuSE. I started with a
fresh full installation of SuSE.

The last lines of stdout and the full stderr are attached below.

Thanks very much for your assistance.

-Ramon F Herrera



stdout:
---
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  
-O6 -march=i686-fomit-frame-pointer-c -o ast_expr2f.o 
ast_expr2f.c
gcc -g  -o asterisk -Wl,-E  io.o sched.o logger.o frame.o loader.o config.o 
channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o 
callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o 
asterisk.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o 
enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o config_old.o plc.o 
jitterbuf.o dnsmgr.o devicestate.o netsock.o slinfactory.o ast_expr2.o 
ast_expr2f.o editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl 
-lpthread -lncurses -lm -lresolv   -lssl
for x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do 
make -C $x || exit 1 ; done
make[1]: Entering directory 
`/home/ramon/ftp/asterisk/1.2/asterisk-1.2.0-beta1/res'
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  
-O6 -march=i686-fomit-frame-pointer -DOPENSSL_NO_KRB5 -fPIC   
-c -o res_adsi.o res_adsi.c
gcc -shared -Xlinker -x -o res_adsi.so res_adsi.o
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  
-O6 -march=i686-fomit-frame-pointer -DOPENSSL_NO_KRB5 -fPIC   
-c -o res_features.o res_features.c
gcc -shared -Xlinker -x -o res_features.so res_features.o
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  
-O6 -march=i686-fomit-frame-pointer -DOPENSSL_NO_KRB5 -fPIC   
-c -o res_crypto.o res_crypto.c
make[1]: Leaving directory 
`/home/ramon/ftp/asterisk/1.2/asterisk-1.2.0-beta1/res'
---

stderr:
/bin/sh: line 1: curl-config: command not found
/bin/sh: line 1: curl-config: command not found
ar: creating libtime.a
ast_expr2f.c:1784: warning: no previous prototype for `ast_yyget_column'
ast_expr2f.c:1860: warning: no previous prototype for `ast_yyset_column'
ast_expr2f.c:1259: warning: `yyunput' defined but not used
res_crypto.c:15:25: openssl/ssl.h: No such file or directory
res_crypto.c:16:25: openssl/err.h: No such file or directory
res_crypto.c:75: error: parse error before RSA
res_crypto.c:75: warning: no semicolon at end of struct or union
res_crypto.c:85: error: parse error before '}' token
res_crypto.c: In function `pw_cb':
res_crypto.c:102: error: dereferencing pointer to incomplete type
res_crypto.c:104: error: dereferencing pointer to incomplete type
res_crypto.c:104: error: dereferencing pointer to incomplete type
res_crypto.c:105: error: dereferencing pointer to incomplete type
res_crypto.c:107: error: dereferencing pointer to incomplete type
res_crypto.c:109: error: dereferencing pointer to incomplete type
res_crypto.c:110: error: dereferencing pointer to incomplete type
res_crypto.c:116: error: dereferencing pointer to incomplete type
res_crypto.c: In function `ast_key_get':
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type
res_crypto.c:127: error: dereferencing pointer to incomplete type

Re: [Asterisk-Users] Voicemail as an email attachement

[Tzafrir Cohen]

On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote:

 I was playing with mta, but this is so complicated, specially if you are on
 dynamic ip address, so it is much easier to use smtp for sending mails..

Sending is never a problem. Recieving is a problem when you're on a
dynamic address.

You can tell your MTA to do just that. e.g, on postfix, in
/etc/postfix/main.cf:

# assuming a well-behaved setup
relayhost = the.isp.domain
# and if not:
relayhost = [smtp.the.isp.domain]

BTW: one option you have with a decent mailer is not to write the email
address in voicemail.conf, but rather, to write there for each box the
email vmbox-vmbox, and use the MTA's aliases to map them to emails.
Either using a plain text /etc/aliases, or using any other database
(ldap, mysql, whatever).

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE: [Asterisk-Users] Asterisk in Croatia - Zagreb

[[EMAIL PROTECTED]]

Hello,

I'm there with you, dude, haven't talked to you in some 5-6 years? :) I
know a couple of people that are working with Asterisk...

Cheers,
Vedran.



mail2web - Check your email from the web at
http://mail2web.com/ .


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Re: [Asterisk-Users] Can't build Asterisk on SuSE

[Dave Cotton]

On Thu, 2005-10-20 at 03:31 -0400, [EMAIL PROTECTED] wrote:
 SuSE Linux Enterprise Server 9
 Asterisk 1.2.0 beta1
 
 I am trying to build and install Asterisk on SuSE. I started with a
 fresh full installation of SuSE.
 
 The last lines of stdout and the full stderr are attached below.
 
 Thanks very much for your assistance.
 
 -Ramon F Herrera

 
 stderr:
 /bin/sh: line 1: curl-config: command not found
 /bin/sh: line 1: curl-config: command not found
 res_crypto.c:15:25: openssl/ssl.h: No such file or directory
 res_crypto.c:16:25: openssl/err.h: No such file or directory

Read the error message. Note what's missing. Install it.


-- 
Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] initiate call recording from phone.

[Tomislav Parcina]

 This (W and w) work for you? Can you tell me can I put both W and w in Dial 
command? You have specified *# in features.conf? Can you tell me how does your 
features.conf looks like?

Tank you for your time!


--
Tomislav Parcina
Lama d.o.o.
www.lama.hr
tparcina#lama.hr 


 Well... I don't know anything about [EMAIL PROTECTED]  I know even more 
 nothing about dialparties.agi... but I can summarize 
 http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
  for you:
 
 Let's say you want to call out on a PSTN line.  A command 
 such as the following will be in your outgoing context:
 exten = x,1,Dial(Zap/2/18005551212,,W)
 before the first comma means dial 18005551212 out the second 
 Zap line, the fact that there's nothing between the 2nd and 
 3rd comma means wait forever for an answer, and the W means 
 let the _calling_ user (you) start a recording (in my case, with *#)
 
 Let's say you want to be able to record incoming calls from 
 PSTN.  A command such as the following would be in your 
 incoming context:
 exten = s,1,Dial(SIP/110,20,w)
 The SIP/110 is where to ring when an incoming call comes in, 
 the 20 means wait 20 seconds before proceeding (to voicemail, 
 or whatever you
 want) and the small w means let the _called_ user (you, 
 again) start a recording however configured.
 
 So... if you don't have direct control over your 
 extensions.conf (as I said, I don't know [EMAIL PROTECTED]) I don't know if 
 you can get your hands dirty with things like this.  Probably 
 there's a check-box in [EMAIL PROTECTED] somewhere that allows this.
 
 good luck!
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[Asterisk-Users] what hardware should I use for asterisk? [please help]

[extreme2000]

Hi,
First of all, I would like to say hello to everybody, it's my first post on the 
list.

I'm building a pbx for a client and I need help/suggestions on what hardware 
and os to choose. I've read all I could find on the net, but still can't decide 
myself. Appart from signal switching, the main concern here is reliability.
The config will stand as follows: 15 sip phone terminals, 4 POTS France telecom 
lines, 1 ISDN line, 4 ip-providers lines, all this will run on on a france 
telecom (argh) dsl line (20M/1M)

In the begining there will be quite a lot of load on this network, but in the 
future the client wishes to connect 30 WAN sip terminals to the asterisk server 
and add 8-10 ip-pstn lines. From what I've heard Asterisk is quite hungry on 
ressources, what kind of hardware can you suggest me to use? Is it worth to buy 
a server mainboard?

And then will the T410P with 4 FXO work together with a T1 (I've heard it was 
not recommended to use 2 digiums on the same M-board).

The second point is about OS, I thought about some free BSD or Solaris and also 
Debian, the first two for quality and Debian because it's well documented and I 
like it, but I don't have any serious opinion on that neither.

Thanks,

Jays

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Re: [Asterisk-Users] New ISDN architecture available for asterisk

[Lenz]

Hi Matteo,
it looks really promising. I'll give it a try!
l.


On Wed, 19 Oct 2005 23:38:00 +0200, Matteo Brancaleoni  
[EMAIL PROTECTED] wrote:




Hi to all,

sorry for crossposting the -dev and -user lists, but I think this could
be quite interesting news for EuroISDN people, expecially BRI owners.

A new ISDN architecture, called vISDN, has been developed to fully
support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and
HFC-8S (with HFC-E1 and HFC-S USB support coming soon).

vISDN is not based on Zaptel, libpri, chan_zap, zaphfc, qozap, etc...
but has been designed from scratch to be a standard compliant EuroISDN
implementation plus a channel crossconnector, plus protocol analisys
support thru Ethereal, plus a ppp terminator, plus other stuff :)




--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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[Asterisk-Users] Any good docs for latest CVS-HEAD / Stable 1.2?

[Sherwood McGowan]

I've been poring 
over the sample configs for the latest CVS-HEAD as well as the readmes from the 
source's docs directory. I'm finding a lot of options that weren't previously 
available, and would like to know if anyone's gone so far as to play with these 
various new settings and document them?

Grateful for any 
help possible...

Ooooh, and 
AEL...what a great idea!!!

SKM
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Re: [Asterisk-Users] Can't build Asterisk on SuSE

[Gerald Dachs]

 SuSE Linux Enterprise Server 9
 Asterisk 1.2.0 beta1

 I am trying to build and install Asterisk on SuSE. I started with a
 fresh full installation of SuSE.

 The last lines of stdout and the full stderr are attached below.

 Thanks very much for your assistance.

 -Ramon F Herrera

[cutted much lines]

 res_crypto.c:15:25: openssl/ssl.h: No such file or directory
 res_crypto.c:16:25: openssl/err.h: No such file or directory
 res_crypto.c:75: error: parse error before RSA

[cutted much lines]

This is my first post to this list, I have no experiences with asterisk,
but this problem is an easy one and it is not asterisk related.
The problem is that you didn' t read the error messages. In the lines above
you can see that you did't install the development files for openssl. I
don't know how this rpm is named in suse, but in my distro it is called
openssl-devel.

Gerald

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Re: [Asterisk-Users] Please recommend a phone

[Omar A. Sabek]

The Cisco CP-7940/60 flashes it's MWI during incoming calls.

If you are using an ATA, there are several devices that can display
flashing/blinking lights during incoming calls by simply putting it
between the ATA and phone.

OmarOn 10/19/05, Christian Stredicke [EMAIL PROTECTED] wrote:
Take a look at snom.com...CS -Original Message- From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Jesse Keating Sent: Wednesday, October 19, 2005 5:31 PM To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Please recommend a phone On Wed, 2005-10-19 at 16:39 -0400, Jesus Mogollon wrote:  I'm in need of a phone that would blink a led to let the callee
  know that there is an incoming call. The GXP-2000 does this but I want  an alternative to Grandstream. Any help is appreciated. Polycom IP301s and 501s have a red LED that blinks when calls
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RE: [Asterisk-Users] Voicemail as an email attachement

[Goran Skular]

On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote:

 I was playing with mta, but this is so complicated, specially if you are
on
 dynamic ip address, so it is much easier to use smtp for sending mails..

Sending is never a problem. Recieving is a problem when you're on a
dynamic address.

You can tell your MTA to do just that. e.g, on postfix, in
/etc/postfix/main.cf:

# assuming a well-behaved setup
relayhost = the.isp.domain
# and if not:
relayhost = [smtp.the.isp.domain]

BTW: one option you have with a decent mailer is not to write the email
address in voicemail.conf, but rather, to write there for each box the
email vmbox-vmbox, and use the MTA's aliases to map them to emails.
Either using a plain text /etc/aliases, or using any other database
(ldap, mysql, whatever).

If relaying is enabled and accepted on remote side... and nowdays is hard to
enable relaying with those spammers around..

I tried something with this relaying, but without success, so I changed
app_voicemail in order to send mail with SMTP and sendEmail script.

Can you tell me how to accept relaying on server, but to limit it to
allowable IP address (which is in this case dynamic ip..).

That will help me a lot :) 

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Re: [Asterisk-Users] Voicemail as an email attachement

[Tzafrir Cohen]

On Thu, Oct 20, 2005 at 08:35:04AM +0200, Goran Skular wrote:
 Small.. just app_voicemail.c and a sendEmail script...
 
 You can download it from here:
 
 app_voicemail.c
 http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f
 ileinfoid=9
 and
 
 
 sendEmail
 http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f
 ileinfoid=10

Project's homepage is
http://caspian.dotconf.net/menu/Software/SendEmail/

(try a google search or a freshmeat search, don't trust my word for it)

Always download programs directly from the homepage or from another
reliable source. Don't just grab programs and scripts from everywhere.

But why not just set mailcmd in voicemail.conf?

Also, quoting the homepage:

 Why not use sendmail?
 Sendmail is a large and complex mail server. Installing this kind of
 mail software on servers (unless it's a mail server) is more of a
 security risk than its worth. 

Not if it only listens on localhost or doesn't listen at all. The
codebases of sendmail is indeed known to be a source of many security
breaches, but exim, postfix and qmail are not so. Most distros come with
either postfix or exim by default nowadays.

 Not to mention it can be a real pain
 messing with configuration files and such. Systems need another simpler
 way to send email from the command prompt, and sendEmail provides this
 functionality. Its a simple, direct way to send email without the
 overhead of other conventional email software.

Most of the pain is caused due to management of messages in the queue.
Other types of pain are due to messages routing. Routing issues can be
easily solved by sending basically all mail to a remote host (excpt,
maybe, some system messages).

However, if the system is disconnected from the net for a while what
will you do? lose all voicemail messages? (and get just ugly warnings 
in the logs as a reminder)

Also note that there are quite a few programs that could use a
sendmail-compatible interface. cron sends its output using mail. So are
many other programs. If you don't provide a sendmail-compatible
interface (even if it one that does not queue, something like
nullmailer) you'll have to reconfigure other parts of your system as
well.

And worst of all: you won't be able to send mail with mutt. The horors!

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE: [Asterisk-Users] Asterisk in Croatia - Zagreb

[Goran Skular]

Hello,

I'm there with you, dude, haven't talked to you in some 5-6 years? :) I
know a couple of people that are working with Asterisk...

Cheers,
Vedran.

Nice surprise ! :)

Ok, you're the first participant along with me on this small gathering. I
sent you email, and let's ring on those guys you know.

I hope that we will find some people out there for a nice gathering on that
subject (and subjects involved in our past 5-6 years you mentioned :) )

See you,
Goran

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RE: [Asterisk-Users] New ISDN architecture available for asterisk

[Goran Skular]

Hi to all,

sorry for crossposting the -dev and -user lists, but I think this could
be quite interesting news for EuroISDN people, expecially BRI owners.

A new ISDN architecture, called vISDN, has been developed to fully
support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and
HFC-8S (with HFC-E1 and HFC-S USB support coming soon).

vISDN is not based on Zaptel, libpri, chan_zap, zaphfc, qozap, etc...
but has been designed from scratch to be a standard compliant EuroISDN
implementation plus a channel crossconnector, plus protocol analisys
support thru Ethereal, plus a ppp terminator, plus other stuff :)


Very, very nice.. I am looking forward for test it.
Further, I hope that ecgo cancelation will be implemented also in near
future, as it is very important in most cases.

Are there maybe some HFC (both BRI and PRI) boards with hw echo cans, or
they are all passive?

For small Euro BRI installations we are using at this moment HFC with
bristuff. But where E1 is involved, we are trying now to avoid E1 cards
without HW echo cans integrated. At this point we are considering between
Sangoma and Digium with hw cans... but who knows what HFC boards would
bring. Beronet and Junghanns are here to be observed..

Kind regards,
Goran

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Re: [Asterisk-Users] chan_capi-0.6 configuration Query with Eicon Diva 4BRI

[Voicomm User]

Okay let me share my experience.

I had 'controller=1,2,3,4' and 'devices=2' in my capi.conf

Devices should be the *sum* of capacity of all controllers i.e in my
case 'devices=8'.

For some reason the exchange didn't like it when I had my controllers
listed over mutiple lines, i.e like john's config. Caller kept getting
the message service not compatible.

Hope this helps.
-r

On 10/20/05, John Daragon [EMAIL PROTECTED] wrote:
 Voicomm User wrote:
  So, are you saying 'msn=' parameter is not required for both Point to
  Point and Point to Multi Point?
 

 Yep.  Dial syntax used to be CAPI/MSN:number so each controller
 needed at least one MSN.

 with 0.6 you can only dial by group, controller or interface name :


 Dial(CAPI/g1/number)
 Dial(CAPI/contr1/number)
 Dial(CAPI/name I gave this controller/number)


 You *will* have to set incomingmsn either to a real number or *.

 jd

 --

 John Daragon  [EMAIL PROTECTED]
 argv[0] limited
 Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
 v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127



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[Asterisk-Users] Isdntrace utility

[Giordano Grandis]

Hi all,

im looking for an utility that let me trace an
ISDN trunk (or all ISDN traffic) on HFC PCI card.



Is there anyone who could help me ?



Any ideas ?



Giordano








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RE: [Asterisk-Users] Voicemail as an email attachement

[Goran Skular]

Always download programs directly from the homepage or from another
reliable source. Don't just grab programs and scripts from everywhere.

But why not just set mailcmd in voicemail.conf?

Also, quoting the homepage:

 Why not use sendmail?
 Sendmail is a large and complex mail server. Installing this kind of
 mail software on servers (unless it's a mail server) is more of a
 security risk than its worth.

Not if it only listens on localhost or doesn't listen at all. The
codebases of sendmail is indeed known to be a source of many security
breaches, but exim, postfix and qmail are not so. Most distros come with
either postfix or exim by default nowadays.

 Not to mention it can be a real pain
 messing with configuration files and such. Systems need another simpler
 way to send email from the command prompt, and sendEmail provides this
 functionality. Its a simple, direct way to send email without the
 overhead of other conventional email software.

Most of the pain is caused due to management of messages in the queue.
Other types of pain are due to messages routing. Routing issues can be
easily solved by sending basically all mail to a remote host (excpt,
maybe, some system messages).

However, if the system is disconnected from the net for a while what
will you do? lose all voicemail messages? (and get just ugly warnings
in the logs as a reminder)

Also note that there are quite a few programs that could use a
sendmail-compatible interface. cron sends its output using mail. So are
many other programs. If you don't provide a sendmail-compatible
interface (even if it one that does not queue, something like
nullmailer) you'll have to reconfigure other parts of your system as
well.

And worst of all: you won't be able to send mail with mutt. The horors!

I completely agree! This is only a work-around.. there are much better
methods involved with sendmail which is really powerfull and thus really
complicated to configure. The most difficult part is not on * server side,
but on relaying server side which must be configured to allow relays only
from authorized sites.. I had no success with that, even with some keys and
similar solutions which I tried, so I gave up and start using sendEmail.

But, I will for sure migrate to sendmail when time for that comes, and I
strongly suggest it to everyone.

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Re: [Asterisk-Users] Any good docs for latest CVS-HEAD / Stable 1.2?

[Olle E. Johansson]

Sherwood McGowan wrote:
 I've been poring over the sample configs for the latest CVS-HEAD as well
 as the readmes from the source's docs directory. I'm finding a lot of
 options that weren't previously available, and would like to know if
 anyone's gone so far as to play with these various new settings and
 document them?
  
 Grateful for any help possible...
  
There's a great book published by O'Reilly called Asterisk -the future
of telephony that covers most of 1.2. It is also available free online
at http://www.asteriskdocs.org

/Olle
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RE: [Asterisk-Users] Asterisk in Croatia - Zagreb

[Goran Skular]

At this moment we are counting 4
possible participants. (Appoligies for those who are not from Croatia for using this list, but this list has a
lot of subscribers including from Croatia)



We are waiting for others to
join us. Feel free to respond here or on my e-mail.



Thanks!









Hello,



I was wondering how many people from Croatia are using and playing with
Asterisk. Recently I had a contact with one user and I am very glad.

It will be really nice to organize a Croatian
Asterisk community and on that way we are organizing a little gathering.

It does not matters how much experience you have,
everthing you need is some interest in Asterisk.

Beside my last contact I know that croatian wifi
community ZG Wireless is using Asterisk also.



So, 



Everyone of you, located in Croatia, please contact me here or
on email.



For the purpose of collecting as much people,
gathering is to be expected next month (around 19th)



Send me an e-mail or even register on www.migo-systems.com. Further info will
be available later.



Looking forward for it,



Goran Skular

www.slsolucije.hr










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[Asterisk-Users] toll free dialing problems using SIP

[Francesco Fondelli]

Hi all,

I have problems when a SIP terminal try to call a toll free number. This
is a call flow that explain what is going on (see comments below and inline):


SIP terminal   Asterisk   NGW  Foo(tool free numb or 
free message)
 ||||
 |  INVITE(SDP)   |||
 |---|  INVITE(SDP)   ||
 ||---||
 |  100   |  100   ||
 |---|---||
 |180(why?)   |||
 |---|||
 ||| IAM|
 |||---|
 ||| ACM|
 ||183(SDP)|---|
 |   no 183 ?!|---||
 ||||
 |||  One Way Voice |
 |||===|
 .
 .
 . RTP data is flowing from bob to Asterisk (checked with tcpdump).
 . RTP data is not forwarded by Asterisk to SIP terminal
 .
 . 30s timeout, SIP terminal keep ringing
 .
 .
 ||||
 ||   CANCEL   ||
 ||---||
 || 200||
 ||---| REL|
 |||---|
 ||| RLC|
 || 487|---|
 ||---||
 || ACK||
 ||---||
 .
 .
 .

1) Why asterisk is sending 180 to SIP terminal? Did I configure * the wrong way?
2) Why 183 with SDP is not forwarded to the SIP terminal?

I have tried canreinvite=[yes|no] and progressinband=[yes|no] and 
pedantic=[yes|no] in
sip.conf but still same behaviour occur. Did I missing something?

Thank you very much, I really need help

Ciao
FF


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Re: [Asterisk-Users] Voicemail as an email attachement

[Tzafrir Cohen]

On Thu, Oct 20, 2005 at 10:06:15AM +0200, Goran Skular wrote:
 On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote:
 
  I was playing with mta, but this is so complicated, specially if you are
 on
  dynamic ip address, so it is much easier to use smtp for sending mails..
 
 Sending is never a problem. Recieving is a problem when you're on a
 dynamic address.
 
 You can tell your MTA to do just that. e.g, on postfix, in
 /etc/postfix/main.cf:
 
 # assuming a well-behaved setup
 relayhost = the.isp.domain
 # and if not:
 relayhost = [smtp.the.isp.domain]
 

 If relaying is enabled and accepted on remote side... and nowdays is hard to
 enable relaying with those spammers around..

If relaying is not allowed then sendEmail won't work as well. Both
senmail (or any other MTA) and sendEmail send their mail by MTA.

Your ISP should relay your SMTP traffic. 

 
 I tried something with this relaying, but without success, so I changed
 app_voicemail in order to send mail with SMTP and sendEmail script.
 
 Can you tell me how to accept relaying on server, but to limit it to
 allowable IP address (which is in this case dynamic ip..).

This is pretty standard configuration of sendmail/postfix/whatever.
Please contact me in private mail (or better: read their docs)

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] more voip patent madness

[Dave Cotton]

On Wed, 2005-10-19 at 12:49 -0700, trixter aka Bret McDanel wrote:
 Teles obtains US patent also for VoIP telephony method
 http://www.heise.de/english/newsticker/news/65126
 
 They are already involved in a lawsuit in germany over their patent
 there, now that they have a US patent expect lawsuits in the US as
 well.  
 
 This just adds to the about 100 patents on VoIP that sprint-nextel
 has, and sprint-nextels willingness to sue.  Course sprint-nextel cant
 do boost mobile services anymore becuase prepaid mobile service is
 patented.  What goes around comes around, and its all insane.

Have a look at this article

http://www.groklaw.net/article.php?story=2005101916522254

Some of the comments are interesting.

With any luck the US will patent itself into a corner. 


-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

[Rich Adamson]

 Darren Thanks for your reply to my problem with the same setup, I have
 found the problem to be Telco related and had it fixed since. But not
 before I tried a Mediatrix 1204 on that setup. It was then that I
 ralized that the problem is with the telco.

Do you know what the specific problem was with the telco?


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Re: [Asterisk-Users] zaptel.conf config for CAS signalling

[Humberto Aicardi]

Even better, share the whole zaptel.conf

Humberto

would you please share line 213 with us?

On 10/18/05, Matt Hess [EMAIL PROTECTED] wrote:
  

I have a customer that needs to do cas signaling across a t1,esf span..
it looks like this can be done but I'm not sure how as the documentation
is very light in regards to cas.. it would appear that I need to use sf
signaling but I get an error saying:
$ ztcfg -vv
Notice: Configuration file is /etc/zaptel.conf
line 213: Unknown keyword 'sf'

I've also tried the format suggested in zaptel.conf

channel# = (etc.)

but I continue to fail.. I'd love a few pointers here..


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Re: [Asterisk-Users] more voip patent madness

[trixter aka Bret McDanel]

On Thu, 2005-10-20 at 11:28 +0200, Dave Cotton wrote:
 Have a look at this article
 
 http://www.groklaw.net/article.php?story=2005101916522254
 
 Some of the comments are interesting.
 
 With any luck the US will patent itself into a corner. 
 
 

Yup its insane, with that ruling it wouldnt be hard for someone to
patent basically the selling of VoIP, forcing all small players under
(or pay a license fee) and the large ones to either negotiate or pay
large sums in court.

This just gives me one more reason to leave, as if I didnt have enough
already.  I am just waiting for my lawsuit to be settled so I can
escape.  It sucks in America, its only getting worse (the patent stuff
is just one more reason) and it wont change untill congress has more
independants elected in.  66% of congress being of like mind and
independant could change a lot of the sillyness like this patent issue,
but you will never see that.  66% can stop any filibuster, 66% can
overrule any veto, 66% can do a lot really quick.  We only need 2 years
of that (the house has 2 year terms).  

As long as the patent office is the way it is though I need to get on to
filing my patent on a time displaced communications system (receive
messages before they are sent, all the components are proven already
just not stable :)


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] more voip patent madness

[Paul]

trixter aka Bret McDanel wrote:


On Thu, 2005-10-20 at 11:28 +0200, Dave Cotton wrote:
 


Have a look at this article

http://www.groklaw.net/article.php?story=2005101916522254

Some of the comments are interesting.

With any luck the US will patent itself into a corner. 



   



Yup its insane, with that ruling it wouldnt be hard for someone to
patent basically the selling of VoIP, forcing all small players under
(or pay a license fee) and the large ones to either negotiate or pay
large sums in court.

This just gives me one more reason to leave, as if I didnt have enough
already.  I am just waiting for my lawsuit to be settled so I can
escape.  It sucks in America, its only getting worse (the patent stuff
is just one more reason) and it wont change untill congress has more
independants elected in.  66% of congress being of like mind and
independant could change a lot of the sillyness like this patent issue,
but you will never see that.  66% can stop any filibuster, 66% can
overrule any veto, 66% can do a lot really quick.  We only need 2 years
of that (the house has 2 year terms).  


As long as the patent office is the way it is though I need to get on to
filing my patent on a time displaced communications system (receive
messages before they are sent, all the components are proven already
just not stable :)

 


I'm going to patent email and web methods for discussing patent issues.

I'm also going to patent methods of emigrating from the US to escape 
inane patent laws.


You can't reply without a license from me.

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Re: [Asterisk-Users] more voip patent madness

[trixter aka Bret McDanel]

On Thu, 2005-10-20 at 06:13 -0400, Paul wrote:
 I'm going to patent email and web methods for discussing patent issues.
 
 I'm also going to patent methods of emigrating from the US to escape 
 inane patent laws.
 
That isnt the primary reason, its just one in a long list :)


 You can't reply without a license from me.


I just filed a disclosure document so I beat you to it, and have 2 years
to acutally patent it.  :P

But seriously with the way the patent stuff is getting, especially what
I read on groklaw, it is entirely possible to patent selling softgoods
like voip.  Small companies cant fight that when so many of those patent
holders are fairly well funded companies that do nothing but file
patents on anything they can get away with.  That makes it really hard
to operate a business with any service in America (afaik no other
government allows such sillyness to that extent).  There is getting to
be a real need to form a trade association to protect the ability to
offer that.  With many of the patents that exist on VoIP technologies
already (do you have enough to fight sprint-nexttel with their 'about
100 patents' ?  What about the German company?).  Some of these patents
are vague enough that combined it would seem that any packetization of
voice onto a data network would qualify.  

Scary thought in my opinion because it makes it such that using asterisk
for non-commerical uses in your own home, either just for fun or as an
answering machine on steroids, can potentially get you into legal hot
water.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] changing the filename of incoming call recordings

[KRTorio]

Is there an easyway to modify the filename of an incoming call's recording, or are we stuck to agent--unix timestamp format given to us by Asterisk?

There seems to beneither anequivalent ChangeMonitor() application for incoming, nor you can tweakthe recording's filenamein agents.conf.
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[Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

[Bartosz Piec]

Hello,

I've got Swissvoice IP10S (SIP) phone and I'm trying it to communicate 
with Asterisk. When I dial from external, the phone rings. But...
On the phone lcd there is a Waiting for proxy server... message all 
the time. Why is it? Phone is set to register in its config. 'sip show 
peers' tells that everithing is ok:


*CLI sip show peers
Name/usernameHostDyn Nat ACL Mask Port 
Status
64   192.168.99.2255.255.255.05060 
OK (60 ms)


I cannot dial from phone to anywhere (busy signal).

Here is my config:
sip.conf as in samples, with added at the end:

[xx]
type=friend
host=192.168.99.2
mask=255.255.255.0
threewaycalling=yes
transfer=yes
singlepath=yes
callwaiting=no
cancallforward=yes
callerid = Sven Svoboda
qualify=yes
username=xx
secret=xx
canreinvite=no
nat=no

extensions.conf as in samples, with added in the [demo] section:
exten = xxx,1,Dial(SIP/xx)

(xxx is the number I calling from)

--
Best regards,
Bartosz Piec
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RE: [Asterisk-Users] Any good docs for latest CVS-HEAD / Stable 1.2?

[Sherwood McGowan]

Thanks for the heads up. I actually have that book, but I'm going to have to
re-read it because I could have sworn things like call-limit and crypto were
not in there before.

I do have to say, however, that the book is phenomonal. I've been running
asterisk in a 1K+ (up to around 3K now) for about 6 months, and it still
showed me some new things I hadn't thought of.

Congrats to O'Reilly for releasing another fine book.

SKM 

--Original Message-
-From: [EMAIL PROTECTED] 
-[mailto:[EMAIL PROTECTED] On Behalf Of 
-Olle E. Johansson
-Sent: Thursday, October 20, 2005 5:12 AM
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: Re: [Asterisk-Users] Any good docs for latest 
-CVS-HEAD / Stable 1.2?
-
-Sherwood McGowan wrote:
- I've been poring over the sample configs for the latest CVS-HEAD as 
- well as the readmes from the source's docs directory. I'm finding a 
- lot of options that weren't previously available, and would like to 
- know if anyone's gone so far as to play with these various new 
- settings and document them?
-  
- Grateful for any help possible...
-  
-There's a great book published by O'Reilly called Asterisk 
--the future of telephony that covers most of 1.2. It is also 
-available free online at http://www.asteriskdocs.org
-
-/Olle
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[Asterisk-Users] Context configuration with AstTapi

[James Steven]

Hi
I am using Asterisk 
TAPI driver with Outlook and have many contacts with numbers listed as +44 1XXX 
XX which is international dialling for UK. My Asterisk context is as 
follows:

[outlook]
exten = 
_0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = 
_00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})

How can I set up the 
context to dial a number starting with +44 from Outlook. I have 
tried:


exten = 
_+.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})

and


exten = 
_+44.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})

but both do not dial 
number. Can Asterisk be set to recognise "+" and change it to 
"00"?

Thanks for your 
help

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Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System

[Vahan Yerkanian]

I'd recommend using native mp3 support that is available in CVS HEAD, as 
madplayer mp3 decoder gives a lower quality sound (audibly more 
cranky/noisy).


Vahan

Jason Becker wrote:

Steve Totaro wrote:


Anyone know how to get around this?  I am stumped.

# make mpg123
[ -f mpg123-0.59r.tar.gz ] || fetch
http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
[ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz
make -C mpg123-0.59r linux
make[1]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
make CC=gcc LDFLAGS= \
OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \
audio_oss.o term.o' \
CFLAGS='-DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX \
-DREAD_MMAP -DOSS -DTERM_CONTROL\
-Wall -O2 -m486 \
-fomit-frame-pointer -funroll-all-loops \
-finline-functions -ffast-math' \
mpg123-make
make[2]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
make[3]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX  
-DREAD_MMAP -DOSS -DTERM_CONTROL-Wall -O2 
-m486 -fomit-f
rame-pointer -funroll-all-loops -finline-functions 
-ffast-ma

th   -c -o mpg123.o mpg123.c
`-m486' is deprecated. Use `-march=i486' or `-mcpu=i486' instead.
cc1: error: CPU you selected does not support x86-64 instruction set
make[3]: *** [mpg123.o] Error 1
make[3]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make[2]: *** [mpg123-make] Error 2
make[2]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make[1]: *** [linux] Error 2
make[1]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make: *** [mpg123] Error 2



Use madplayer instead. There are several reasons why Digium  the 
Asterisk community should part ways with mpg123.


Regards,


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[Asterisk-Users] Siwssvoice IP10S telnet password

[Bartosz Piec]

Hello,

Does anyone know what is the default password for telnet in Swissvoice 
IP10S phone? I didn't find any in documentation...


--
Best regards,
Bartosz Piec
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Re: [Asterisk-Users] Siwssvoice IP10S telnet password

[Igor Briski]

Bartosz Piec wrote:

Hello,

Does anyone know what is the default password for telnet in Swissvoice 
IP10S phone? I didn't find any in documentation...


U: target
P: password

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Re: [Asterisk-Users] Siwssvoice IP10S telnet password

[Karim AMER]

username: target
password: password


Hello,

Does anyone know what is the default password for telnet in Swissvoice 
IP10S phone? I didn't find any in documentation...


--
Best regards,
Bartosz Piec
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Cordialement,
__

Karim Amer  IC TELECOM / IC CENTREX
45 quai de Seine 75019 Paris
Direct IP : 01 72 74 82 84
http://www.iccentrex.com

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Re: [Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

[Arnaud Bled]

hi

Which Firmware Version is loaded on the SwissVoice ?
Because only the latest version are RFC3261 based
i can send you offlist a 1.0.0 build Version



//arnaud







At 12:35 20/10/2005, you wrote:

Hello,

I've got Swissvoice IP10S (SIP) phone and I'm trying it to communicate 
with Asterisk. When I dial from external, the phone rings. But...
On the phone lcd there is a Waiting for proxy server... message all the 
time. Why is it? Phone is set to register in its config. 'sip show peers' 
tells that everithing is ok:


*CLI sip show peers
Name/usernameHostDyn Nat ACL Mask Port Status
64   192.168.99.2255.255.255.05060 OK (60 ms)

I cannot dial from phone to anywhere (busy signal).

Here is my config:
sip.conf as in samples, with added at the end:

[xx]
type=friend
host=192.168.99.2
mask=255.255.255.0
threewaycalling=yes
transfer=yes
singlepath=yes
callwaiting=no
cancallforward=yes
callerid = Sven Svoboda
qualify=yes
username=xx
secret=xx
canreinvite=no
nat=no

extensions.conf as in samples, with added in the [demo] section:
exten = xxx,1,Dial(SIP/xx)

(xxx is the number I calling from)

--
Best regards,
Bartosz Piec
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--



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[Asterisk-Users] Why Asterisk documentation is so poor...

[Sergey Okhapkin]

http://bugs.digium.com/view.php?id=5472

The users will not learn about undocumented AEL features. Sure I'm not
going to reopen the problem.

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[Asterisk-Users] wm_w DTMF solution for T1 tie line losing deigits.

[Steven]

I assume the real fix is to alter some DTMF setting in my Panasonic DBS576, 
but I have yet to find it.

I was using a PRI card in my panasonic, but it broke, so I switched to a 
spare T1 card.
I set it up for em_w, but asterisk was dialing before it recieved all of the 
digits.

I saw a few suggestions in the WIKI and mailing list, but none worked as is.

The issue that complicated the exaples the most was the fact that sometimes 
I would recieve 1 digit and sometimes 4 or 6 etc.
If dialed fast enough, I would get the whole number in the fist pass.

[panasonic-catchall] is included last because it is the catchall for all non 
found numbers.
I am using this T1 for both 4 digit extension and as a trunk in the 
panasonic, so I do not have my 9 to route with.

exten = _X, is catching if only 1 digit is passed.
exten = _X., is catching if it is more than one.
exten = _X,5,GotoIf($[${Predigits1} = ]?s-gathermoredigits,1) ; this 
was the trick to make sure I didn't loop from the WaitExten() .

Here is the solution that I found that works 100% for me:


---
[panasonic]

include = ext-local
include = outbound-allroutes
; include = outrt-005-tollfree
; include = outrt-004-dial911
; include = outrt-003-dial9
; include = outrt-002-fwd
include = panasonic-catchall

[panasonic-catchall]

exten = _1X.,2,Dial(Zap/g0/${EXTEN},,r)
exten = _1X.,3,Congestion

exten = _X,1,NoOp( only got a few digit. It was ${EXTEN})
exten = _X,2,SetVar(Predigits1=${Predigits2})
exten = _X,3,SetVar(Predigits2=${EXTEN})
exten = _X,4,GotoIf($[${Predigits1} = ]?s-gathermoredigits,1)
exten = _X,5,NoOp(${TIMESTAMP} ok, now we're going to dial 
${Predigits1}${Predigits2}${EXTEN})
exten = _X,6,Dial(Zap/g0/${Predigits1}${Predigits2}${EXTEN},,r)
exten = _X,7,Congestion

exten = _X.,1,NoOp( only got a few digit. It was ${EXTEN})
exten = _X.,2,SetVar(Predigits1=${Predigits2})
exten = _X.,3,SetVar(Predigits2=${EXTEN})
exten = _X.,4,GotoIf($[${Predigits1} = ]?s-gathermoredigits,1)
exten = _X.,5,NoOp(${TIMESTAMP} ok, now we're going to dial 
${Predigits1}${Predigits2}${EXTEN})
exten = _X.,6,Dial(Zap/g0/${Predigits1}${Predigits2}${EXTEN},,r)
exten = _X.,7,Congestion

exten = t,1,NoOp( timed out dialing ${Predigits1}${Predigits2})
exten = t,2,Dial(Zap/g0/${Predigits1}${Predigits2},,r)
exten = t,3,Congestion

exten = s-gathermoredigits,1,NoOp( users have slow fingers - lets increase 
the DigitTimeout and try again)
exten = s-gathermoredigits,2,DigitTimeout,5; Increase the 'finished 
dialing' timeout to 5 seconds
exten = s-gathermoredigits,3,WaitExten(4)  ; and give the caller 8 
seconds overall to do their thing



-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of 
having a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - 
 - --- - - -- -  -- --   -   -- 



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RE: [Asterisk-Users] Asterisk in Croatia - Zagreb

[ua]

On Čet, 2005-10-20 at 11:17 +0200, Goran Skular wrote:
 We are waiting for others to join us. Feel free to respond here or on
 my e-mail.

Count me in, too.

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[Asterisk-Users] codec voice quality ratings

[trixter aka Bret McDanel]

Has anyone done any quality measurements on the codecs as implemented in
asterisk?  Specifically something along the lines of:
MOS (mean opinion score) either 1-5 or 1-10 variant
DAM (diagnostic acceptability measure)
DRT (diagnostic rhyme test)

Obviously MOS is the easiest, and network, speaker and microphone
quality can affect the results, but I would be interested in seeing
something done which goes into detail about the equipment used in the
test and the sample pool used to test it.

Anyone have any references for this?

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] Asterisk Compilation with H323 working on it

[Carlos Arnt]

Hi Folks,

Can recomend a asterisk compilation for Mandrake or Debian that has on it H323 WORKING ?

I try use H323 with Asterisk for some implementations but that cant good results.

So any tip ?

Thanks alot !

Carlos.



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[Asterisk-Users] Help on Asterisk and Client SIP setup

[Chrispen Chisvo]

hi

I have instaled Asterisk PBX on Linux SUSE.

Its is running well. 

I want to add extensions for a simple test. I have added the extensions like

add extension 137,1,Dial,IAX/192.168.1.37/137 into local   

what I am not clear of is IAX?

And my extensions are failing to register.

Unfortunaltely a client lite xlite will require me to specify username, 
password, and domain, and yet I am not clear where to define the user and the 
domain on the Asterisk PBX.

Anyone to help me setup an extension and the VOIP SIP client xlite: simple 
steps please.

-- 
Rgds
Chrispen Chisvo
Ecoweb Zimbabwe
Cell: +263 91 222 443
Tel: +263 4 758 194
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Re: [Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

[Bartosz Piec]

Arnaud Bled napisał(a):

Which Firmware Version is loaded on the SwissVoice ?


Application version: IP10SP v1.0.0 (Build 11)
Boot version: IP10 Boot v1.0.7
DSP version: Rel9.1.30.6,p8
(what's DSP in fact?)


Because only the latest version are RFC3261 based


So the phone must be RFC3261 compliant to be used with Asterisk, right?


i can send you offlist a 1.0.0 build Version


Would be very appreciated...

--
Best regards,
Bartosz Piec
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[Asterisk-Users] Re: Please recommend a phone

[Doug Meredith]

Jesus Mogollon [EMAIL PROTECTED] wrote:

I'm in need of a phone that would blink a led to let the callee know that
there is an incoming call. The GXP-2000 does this but I want an alternative
to Grandstream. Any help is appreciated.

The Aastra 480i does this.

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
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Re: [Asterisk-Users] Help on Asterisk and Client SIP setup

[Bartosz Piec]

Chrispen Chisvo napisał(a):

I want to add extensions for a simple test. I have added the extensions like

add extension 137,1,Dial,IAX/192.168.1.37/137 into local   


what I am not clear of is IAX?


Are you sure you are using IAX? Below you are writing about SIP...

Unfortunaltely a client lite xlite will require me to specify username, 
password, and domain, and yet I am not clear where to define the user and the 
domain on the Asterisk PBX.


Read about sip.conf and extensions.conf. This is good place to start: 
http://www.voip-info.org/wiki-Asterisk


--
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Bartosz Piec
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RE: [Asterisk-Users] Asterisk Compilation with H323 working on it

[Bohuslav Coufal]

I did use it on Debian and now use it on
FC4 and H323 is working good on both systems. Im using asterisk own h323
driver.



Bob.











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt
Sent: Thursday, October 20, 2005
2:24 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk
Compilation with H323 working on it








Hi Folks,











Can recomend a asterisk compilation for Mandrake or Debian
that has on it H323 WORKING ?











I try use H323 with Asterisk for some implementations but
that cant good results.











So any tip ?











Thanks alot !











Carlos.














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[Asterisk-Users] E1 (TE405p) SetCallerId problem

[Derek Conniffe]

Hi all,

I'm having a problem with some E1 lines I have from the Telco.  They 
have told me that I should be able to specify the outgoing caller id in 
an AREA + NUMBER format (e.g. 14401806 for my Irish number +353 1 440 
1806) but this does not appear to work for me (and either does any other 
apparent combinations of caller Id that I've tried.  The outgoing number 
is just defaulting to one phone number assigned to the E1 line.


Is there any settings relating to Setting the caller id in 
/etc/asterisk/zapata.conf or /etc/zaptel.conf or do you think this is a 
Telco problem?  (I'm holding off complaining yet again to the Telco 
until I'm sure the problem isn't mine!.


I have usecallerid=yes and hidecallerid=no set in 
/etc/asterisk/zapata.conf


I have no problem setting the caller Id and using a VOIP termination 
company to set the caller id - the problem is only with my E1 lines.


Thanks for any help!

Derek

--
Derek Conniffe
Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
United Kingdom: 0870 068 2368
International: 00 353 1 440 1800
Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823)
Fax: 01 201 0085 (International: 00 353 1 201 0085)
Email: [EMAIL PROTECTED]
Web: http://www.rivertowerhosting.com

begin:vcard
fn:Derek Conniffe
n:Conniffe;Derek
org:Rivertower Ltd;IT
adr:Dublin 2;;46 Upper Mount Street;Dublin;Dublin;Dublin 2;Ireland
email;internet:[EMAIL PROTECTED]
tel;work:+353 1 201 0146
tel;fax:+353 1 201 0085
tel;cell:+353 86 856 3823
note;quoted-printable:Ireland: (Freephone) 1800 719 400=0D=0A=
	Ireland: (Local) 01 244 9719=0D=0A=
	United Kingdom: 0870 068 2368=0D=0A=
	International: 00 353 1 244 9719=0D=0A=
	
url:http://www.rivertowerhosting.com
version:2.1
end:vcard

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Re: [Asterisk-Users] DID setup from goiax.com

[Faris Raouf]

trixter aka Bret McDanel wrote:

I dont know then that was cut and paste from what I have working ...

maybe actual log dumps of the error?

On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote:

That is What I stated in the email.. my GOIAX #. not the DID #.

That is not the issue.





Is this still ongoing? If so...

when you get an error like [EMAIL PROTECTED] in the log, it is a good indication 
that something is looking for a priority s in the context (I think).


In my case I set up goiax yesterday and had this exact error. The 
solution was simply to have s priorities in the context in 
extensions.conf that my context in iax.conf was pointing to for goiax.


NOTE: In the following, mygoiaxnumber should be replaced with the 
actual number (not DID number) that you see on your screen when you 
first register, just above you password.



iax.conf:

register = mygoiaxnumber:[EMAIL PROTECTED]

[mygoiaxnumber]
context=goiaxinwards
etc
etc


extensions.conf:

[goiaxinwards]
exten = s,1, Answer()
etc

AND NOT:

exten = mygoiaxnumber,1,Answer()
etc

(which is what I originally had and which did not work for me in my 
particular case - I got the [EMAIL PROTECTED] type error)


Faris.


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[Asterisk-Users] Call Transfer

[Rhonda Herron]

Hello,

I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I 
am testing extended functions for my office users and am hitting a wall. 
I simply need to be able to put a call on hold and forward it to any 
another internal extension. I have an Eezee AT-320 IAX2 phone and 
according to the directions, I  simply select Hold  enter ext hit Fwd. 
However when I press the button all I do is annoy the caller with loud 
button punching sounds. Does something need to be configured in * to 
allow call transfer to work? I am using an inbound trunk from Teliax- no 
cards, just a T1 direct to my * server.  I have found transfer functions 
for zapatel- but as I said I am just using the T1 and have no zapatel 
trunks/configurations.  I have also seen a lot of information for call 
forwarding but that sets up a permanent forward function to a specific 
extension. I just want to be able to say One moment, Mike can help you 
with that, let me transfer you and then be able to do it. Since this 
happens with all my AT-320 phones, I don't think it is hardware related 
and there is no mention of call transfer configuration for the phone 
itself.


Thanks

-R
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Re: [Asterisk-Users] Call Transfer

[Tom Vile]

try # and then dial the extension.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote:
Hello,I have my [EMAIL PROTECTED] working beautifully for basic call function. So now Iam testing extended functions for my office users and am hitting a wall.I simply need to be able to put a call on hold and forward it to any
another internal extension. I have an Eezee AT-320 IAX2 phone andaccording to the directions, Isimply select Hold  enter ext hit Fwd.However when I press the button all I do is annoy the caller with loud
button punching sounds. Does something need to be configured in * toallow call transfer to work? I am using an inbound trunk from Teliax- nocards, just a T1 direct to my * server.I have found transfer functions
for zapatel- but as I said I am just using the T1 and have no zapateltrunks/configurations.I have also seen a lot of information for callforwarding but that sets up a permanent forward function to a specific
extension. I just want to be able to say One moment, Mike can help youwith that, let me transfer you and then be able to do it. Since thishappens with all my AT-320 phones, I don't think it is hardware related
and there is no mention of call transfer configuration for the phoneitself.Thanks-R___--Bandwidth and Colocation sponsored by 
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To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205
Phone: 978-203-3848 x205Fax: 518-631-2856
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Re: [Asterisk-Users] Asterisk in Croatia - Zagreb

[Igor Briski]

ua wrote:


We are waiting for others to join us. Feel free to respond here or on
my e-mail.


Count me in, too.


We should probably open a separate mailing list for croatian users. Much 
easier to communicate and we avoid clogging up this list.


Nice to see that there are so many asterisk users here in Croatia. :)

--
Igor Briski -- [EMAIL PROTECTED]
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RE: [Asterisk-Users] wm_w DTMF solution for T1 tie line losing deigits.

[Dennis Walker]

I had a similar problem with a wink tie t1
try setting the emdigitwait=[ms]   in zapata.conf

on my system  I set  emdigitwait=600

--
From:   Steven[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Thursday, October 20, 2005 8:17 AM
To: asterisk-users@lists.digium.com
Subject:[Asterisk-Users] wm_w DTMF solution for T1 tie line losing 
deigits.

I assume the real fix is to alter some DTMF setting in my Panasonic DBS576, 
but I have yet to find it.

I was using a PRI card in my panasonic, but it broke, so I switched to a 
spare T1 card.
I set it up for em_w, but asterisk was dialing before it recieved all of the 
digits.

I saw a few suggestions in the WIKI and mailing list, but none worked as is.

The issue that complicated the exaples the most was the fact that sometimes 
I would recieve 1 digit and sometimes 4 or 6 etc.
If dialed fast enough, I would get the whole number in the fist pass.

[panasonic-catchall] is included last because it is the catchall for all non 
found numbers.
I am using this T1 for both 4 digit extension and as a trunk in the 
panasonic, so I do not have my 9 to route with.

exten = _X, is catching if only 1 digit is passed.
exten = _X., is catching if it is more than one.
exten = _X,5,GotoIf($[${Predigits1} = ]?s-gathermoredigits,1) ; this 
was the trick to make sure I didn't loop from the WaitExten() .

Here is the solution that I found that works 100% for me:


---
[panasonic]

include = ext-local
include = outbound-allroutes
; include = outrt-005-tollfree
; include = outrt-004-dial911
; include = outrt-003-dial9
; include = outrt-002-fwd
include = panasonic-catchall

[panasonic-catchall]

exten = _1X.,2,Dial(Zap/g0/${EXTEN},,r)
exten = _1X.,3,Congestion

exten = _X,1,NoOp( only got a few digit. It was ${EXTEN})
exten = _X,2,SetVar(Predigits1=${Predigits2})
exten = _X,3,SetVar(Predigits2=${EXTEN})
exten = _X,4,GotoIf($[${Predigits1} = ]?s-gathermoredigits,1)
exten = _X,5,NoOp(${TIMESTAMP} ok, now we're going to dial 
${Predigits1}${Predigits2}${EXTEN})
exten = _X,6,Dial(Zap/g0/${Predigits1}${Predigits2}${EXTEN},,r)
exten = _X,7,Congestion

exten = _X.,1,NoOp( only got a few digit. It was ${EXTEN})
exten = _X.,2,SetVar(Predigits1=${Predigits2})
exten = _X.,3,SetVar(Predigits2=${EXTEN})
exten = _X.,4,GotoIf($[${Predigits1} = ]?s-gathermoredigits,1)
exten = _X.,5,NoOp(${TIMESTAMP} ok, now we're going to dial 
${Predigits1}${Predigits2}${EXTEN})
exten = _X.,6,Dial(Zap/g0/${Predigits1}${Predigits2}${EXTEN},,r)
exten = _X.,7,Congestion

exten = t,1,NoOp( timed out dialing ${Predigits1}${Predigits2})
exten = t,2,Dial(Zap/g0/${Predigits1}${Predigits2},,r)
exten = t,3,Congestion

exten = s-gathermoredigits,1,NoOp( users have slow fingers - lets increase 
the DigitTimeout and try again)
exten = s-gathermoredigits,2,DigitTimeout,5; Increase the 'finished 
dialing' timeout to 5 seconds
exten = s-gathermoredigits,3,WaitExten(4)  ; and give the caller 8 
seconds overall to do their thing



-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of 
having a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - 
 - --- - - -- -  -- --   -   -- 



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Fwd: [Asterisk-Users] Re: [Asterisk-doc] You ASKED for an Asterisk book, you GOT an Asterisk book!

[Jerry Richmond]

Note: forwarded message attached.---BeginMessage---
On 10/15/05, Sean Wheller [EMAIL PROTECTED] wrote:
 On Saturday 15 October 2005 20:58, Leif Madsen wrote:
  Asterisk: The Future of Telephony is now freely available, for
  download in PDF form, from the Asterisk Documentation Project website
  located at http://www.asteriskdocs.org.

 Congratulations on the release of this book. It certainly is a great body of
 work.

 I would like to ask whether the authors and O'Reilly Media would ever release
 under a less restrictive license perhaps
 http://creativecommons.org/licenses/by/2.0/ ? Of course I would like cc-by-sa
 2.5, but that may be pushing it :-)

 If the above is a possability, it would enable the source of the book to be
 made available for contribution in the Asterisk docs repository.

In the future, it *may* be a possibility, but for now, commercial
distribution and derivitive works (changes) are *not* allowed at this
time. I believe these are fair requests because from a knowledge
standpoint, the community gains a tremendous amount by having freely
available, professionally edited text, and O'Reilly still retains
control of the work and distribution of the work so that it can first,
pay for the costs involved in the creation and publication of the
book, and second, to make it profitable so as to allow them to
continue creating more great books for everyone.

--
Leif Madsen - http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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RE: [Asterisk-Users] Asterisk Compilation with H323 working on it

[Carlos Arnt]

Hi

Did it work well with Netmeeting from Microsoft ??

Thanks for answer.

Carlos.

On Thu, 20 Oct 2005 14:41:38 +0200, Bohuslav Coufal wrote: I did use it on Debian and now use it on FC4 and H323 is working good on both systems. I’m using asterisk own h323 driver. Bob. From: [EMAIL PROTECTED] [mailto:asterisk-[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Thursday, October 20, 2005 2:24 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Compilation with H323 working on it Hi Folks, Can recomend a asterisk compilation for Mandrake or Debian that has on it H323 WORKING ? I try use H323 with Asterisk for some implementations but that cant good results. So any tip ? Thanks alot ! Carlos.Carlos Arnt
Key soluções em Internet
Av. das americas 500 bl 03 sala 204
Tel: (021) 2492-1666
Voip rede mundial: 9000 ou 9500
E-mail: [EMAIL PROTECTED]



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[Asterisk-Users] user name

[Jerry Richmond]

I am geting e-mail but asterisk doesn't know my user name or password. My user name has always Been Jerry Richmond, my e-mail address [EMAIL PROTECTED] I need a password of some kind.
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Re: [Asterisk-Users] Why Asterisk documentation is so poor...

[Olle E. Johansson]

Sergey Okhapkin wrote:
 http://bugs.digium.com/view.php?id=5472
 
 The users will not learn about undocumented AEL features. Sure I'm not
 going to reopen the problem.

Sergey,
I am sorry if you took our comments that badly. I proposed a worthing
and you did not accept that and refused to update according to our
suggestions. Tilghman therefor decided to close the bug.

I suggest you try again, re-open the bug, fix the problem and continue
to add more documentation. We do need more documentation! It has to be
correct though, and that's why we are giving feedback.

/Olle
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RE: [Asterisk-Users] Asterisk Compilation with H323 working on it

[Bohuslav Coufal]

I dont use Microsoft Netmeeting. Sorry
I use HW H323 devices only. AVAYA S8300 and some Planet telephones.



Bob.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt
Sent: Thursday, October 20, 2005
3:43 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Asterisk Compilation with H323 working on it







Hi











Did it work well with Netmeeting from Microsoft ??











Thanks for answer.











Carlos.











On Thu, 20 Oct 2005 14:41:38 +0200, Bohuslav Coufal wrote:
I did use it on Debian and
now use it on FC4 and H323 is working
good on both systems. I’m
using asterisk own h323 driver.

Bob.


From:
[EMAIL PROTECTED] [mailto:asterisk-
[EMAIL PROTECTED]
On Behalf Of Carlos Arnt
Sent: Thursday, October 20,
2005 2:24 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Asterisk Compilation with H323 working on
it


Hi Folks,


Can recomend a asterisk
compilation for Mandrake or Debian that has
on it H323 WORKING ?


I try use H323 with
Asterisk for some implementations but that cant
good results.


So any tip ?


Thanks alot !


Carlos.


Carlos Arnt





Key soluçőes em Internet





Av. das americas 500 bl 03 sala 204





Tel: (021) 2492-1666





Voip rede mundial: 9000 ou 9500





E-mail: [EMAIL PROTECTED]














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[Asterisk-Users] Asterisk Billing

[Kanishka Somaratne]

Hi
I am looking for a asterisk billing system with a reseller module. for 
example, i there are 2 accoutns admin 1 and admin 2.
when they login only the accounts they created should be shown. admin 2s 
accounts pr rates should not be shown to admin 2.


does astbill support this. please let me know

regards
Kani 


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RE: [Asterisk-Users] user name

[Jonathan k. Creasy]

I dont get it. 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Richmond
Sent: Thursday, October 20, 2005
9:46 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] user
name







I am geting e-mail but asterisk doesn't know my user name or password.
My user name has always Been Jerry Richmond, my e-mail address [EMAIL PROTECTED] I
need a password of some kind.





thanks








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Re: [Asterisk-Users] Asterisk Billing

[Darren Wiebe]

I don't know if astbill supports this or not.  ASTPP does supports it 
though.   www.aleph-com.net/astpp   You would set admin 1 and admin2 up 
as resellers.


Darren Wiebe
[EMAIL PROTECTED]

Kanishka Somaratne wrote:


Hi
I am looking for a asterisk billing system with a reseller module. for 
example, i there are 2 accoutns admin 1 and admin 2.
when they login only the accounts they created should be shown. admin 
2s accounts pr rates should not be shown to admin 2.


does astbill support this. please let me know

regards
Kani
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Re: [Asterisk-Users] Why Asterisk documentation is so poor...

[Sergey Okhapkin]

I made a lot of contributions to many open source projects already, I
never saw such pressure from the code maintainers to code contributors,
usually it's up to maintainers how to apply the changes proposed by the
contributor. I put a note that you can rephrase as you wish to follow
asterisk's maintainers roadmap and guidelines. I'm not fluent in english
also, to express your wishes in the way you want.

On Thu, 2005-10-20 at 15:50 +0200, Olle E. Johansson wrote:
 Sergey Okhapkin wrote:
  http://bugs.digium.com/view.php?id=5472
  
  The users will not learn about undocumented AEL features. Sure I'm not
  going to reopen the problem.
 
 Sergey,
 I am sorry if you took our comments that badly. I proposed a worthing
 and you did not accept that and refused to update according to our
 suggestions. Tilghman therefor decided to close the bug.
 
 I suggest you try again, re-open the bug, fix the problem and continue
 to add more documentation. We do need more documentation! It has to be
 correct though, and that's why we are giving feedback.
 
 /Olle
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Re: [Asterisk-Users] Help on Asterisk and Client SIP setup

[Chrispen Chisvo]

I want to use SIP.

So I want to configure a SIP Xlite to register onto the PBX.

whats are the steps to:
- add an extension for sip in the asterisk PBX when I have an Xlite extension 
with the following configurations:
- username: user1
- authorised user: chris
- password:
- Domain/Realm: 192.168.1.37
- SIP Proxy: 192.168.1.37
- Out Bound Proxy: 192.168.1.37

What would be extension add syntax?

rgds
CC

On Thursday 20 October 2005 06:39, Bartosz Piec wrote:
 Chrispen Chisvo napisał(a):
  I want to add extensions for a simple test. I have added the extensions
  like
 
  add extension 137,1,Dial,IAX/192.168.1.37/137 into local
 
  what I am not clear of is IAX?

 Are you sure you are using IAX? Below you are writing about SIP...

  Unfortunaltely a client lite xlite will require me to specify username,
  password, and domain, and yet I am not clear where to define the user and
  the domain on the Asterisk PBX.

 Read about sip.conf and extensions.conf. This is good place to start:
 http://www.voip-info.org/wiki-Asterisk

-- 
Rgds
Chrispen Chisvo
Ecoweb Zimbabwe
Cell: +263 91 222 443
Tel: +263 4 758 194
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[Asterisk-Users] Manager API - Supervised Transfer

[Richard Cook]

Does anyone have a sample on how to do a 
supervised transfer via the Manager API.

Incoming Zap - SIP - xfer - 
Zap

--Richard Cook[EMAIL PROTECTED]T: 705-223-2000 
x2010
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[Asterisk-Users] Getting output from agi scripts (python)

[Simone Cittadini]

I don't get output in the cli from agi scripts when connecting to a 
running instance of asterisk.

And that is all well and known :
This is a known problem. Asterisk will only send STDERR from AGI 
scripts to the actual console Asterisk is running on

I can't, don't want, to do the
/usr/bin/screen -L -d -m -S asterisk /usr/sbin/asterisk -vgc
trick

So I putted in my python scripts some logging to file, it doesn't work.

logger = logging.getLogger()
logger.setLevel(logging.DEBUG)
hdlr = logging.FileHandler(agi_log.txt)
logger.addHandler(hdlr)
logger.debug(foobar)
hdlr.flush()
hdlr.close()

writes foobar in a file when called from shell, just creates the file if 
integrated in a agi.

(I can't understand how It's a minor issue for most people. btw)

suggestions ? tricks ?
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Re: [Asterisk-Users] Any good docs for latest CVS-HEAD / Stable 1.2?

[Bruce Ferrell]

Olle E. Johansson wrote:

Sherwood McGowan wrote:


I've been poring over the sample configs for the latest CVS-HEAD as well
as the readmes from the source's docs directory. I'm finding a lot of
options that weren't previously available, and would like to know if
anyone's gone so far as to play with these various new settings and
document them?

Grateful for any help possible...



There's a great book published by O'Reilly called Asterisk -the future
of telephony that covers most of 1.2. It is also available free online
at http://www.asteriskdocs.org

/Olle


Bought it.  I have to say it's a wee touch on the thin side.

A lot of pages. Light on theory/principals of asterisk configuration.

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RE: [Asterisk-Users] initiate call recording from phone.

[james.texter]

I'm curious if anyone has this working with [EMAIL PROTECTED]  I just installed 
the 2.0 Beta, which loads up * v1.2.0.  I edited my features.conf to put in the 
following:

[featuremap]
automon = *1

I place a call to my cell phone, and from my polycom put in *1, but nothing 
happens.  If I use [EMAIL PROTECTED] to setup to always record, it works fine.  
If anyone has this working with [EMAIL PROTECTED] 2.0 Beta, please let me know.

Thanks,

James

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan 
 Company, LLC
Sent: Wednesday, October 19, 2005 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] initiate call recording from phone.

Well... I don't know anything about [EMAIL PROTECTED]  I know even more nothing 
about 
dialparties.agi... but I can summarize 
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial for you:

Let's say you want to call out on a PSTN line.  A command such as the 
following will be in your outgoing context:
exten = x,1,Dial(Zap/2/18005551212,,W)
before the first comma means dial 18005551212 out the second Zap line, 
the fact that there's nothing between the 2nd and 3rd comma means wait 
forever for an answer, and the W means let the _calling_ user (you) 
start a recording (in my case, with *#)

Let's say you want to be able to record incoming calls from PSTN.  A 
command such as the following would be in your incoming context:
exten = s,1,Dial(SIP/110,20,w)
The SIP/110 is where to ring when an incoming call comes in, the 20 
means wait 20 seconds before proceeding (to voicemail, or whatever you 
want) and the small w means let the _called_ user (you, again) start a 
recording however configured.

So... if you don't have direct control over your extensions.conf (as 
I said, I don't know [EMAIL PROTECTED]) I don't know if you can get your hands 
dirty 
with things like this.  Probably there's a check-box in [EMAIL PROTECTED] 
somewhere 
that allows this.

good luck!



todd wrote:
 Moj
 First great to see someone has figured this out, I have been struggling with 
 it.
 If not to much trouble; could you spare an example of where that w or W 
 exist in the Dial command. Also will this command in the Dial plan work if I 
 am using [EMAIL PROTECTED]
 And how does this work into the whole picture with the dialparties.agi 
 script, if at all?
 Obviously I am a little confused on how this all works any help would be 
 GREATLY appreciated.
 Todd
 - Original Message - 
 From: Mojo with Horan  Company, LLC [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, October 17, 2005 10:56 AM
 Subject: Re: [Asterisk-Users] initiate call recording from phone.
 
 
 
And the w or W options must be specified in the Dial() cmd, as in:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial

Moj

Mojo with Horan  Company, LLC wrote:

If you have httpd with php on the * server, you can do what I did:

I set up the key combination *# in features.conf to monitor and created a 
few php files to interact with the results.  Save the four php files at:

http://horanappraisals.com/asterisk/

into a folder on the * web server, eg: /var/www/html/recordings/ -- 
rename them all to .php instead of .phps, and edit config.php to point to 
the asterisk monitor directory (usually /var/spool/asterisk/monitor). Now 
make a soft link so the recorded waves appear in the web tree:

ln -s /var/spool/asterisk/monitor /var/www/html/recordings/monitor

Then direct a web browser to http://asterisk_server/recordings/ and it 
should be pretty self-explanatory.  No recordings will appear in the list 
if you don't have the sox packages installed.

Andy Goss wrote:


I am looking for a way to allow a user to record a call simply by
pressing a button or some combination of buttons on their phone.  Is
this possible?

I have read the stuff about the monitor command; however, this doesn't
seem to be very interactive for the user.
Thanks,
Andy

--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]

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-- 
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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-- 
Mojo [EMAIL PROTECTED]
Office Manger, Horan 

[Asterisk-Users] TDMoE and Badness in Kernel

[astgroups]

I'm going to poll the group one more time on this one. I have posted
this before and didn't get any takers. 

Digium advises that I should just do IAX in place of TDMoE but I don't
have that luxury. I have a very complex dial plan built around the TDMoE
functionality and it would be very difficult/expensive to rewrite it.
This has always worked excellent on 2.4 but now that we need to upgrade
to 2.6 I'm getting all kinds of headaches. I'm willing to pay a
consultant to work this out for me. Please contact me off list if
interested

The following is my original message:

Badness in local_bh_enable at kernel/softirq.c on 2.6.X

I'm seeing this on Kernel 2.6.+ implementations, namely Centos 4.1, FC4
machines while trying to do TDMoE trunks between two machines. 
2.4 Kernel operates fine on the same hardware

I'm compiling zaptel-1.0.9.2 as per instructions in README.Linux26 +
README.udev. I've also tried CVS head zaptel.

Here are some references where the issue has been reported before but
I've yet to find a documented solution;

http://lists.digium.com/pipermail/asterisk-users/2005-February/091867.html

http://bugs.digium.com/view.php?id=5126

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[Asterisk-Users] CallerID PHP Script

[David Choo]

Hi All,

As far as I'm aware, there is this PHP
Script that allows us to add / remove callerID from Asterisk's Database?
However, as my HDD crashed, I'm unable to search back my old archives.
Would anyone be kind enough to point me to the correct URL? Thanks.

Best Regards,

==
David Choo
Sales Engineer
Business  Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-6842 2725, Ext - 404
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=

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[Asterisk-Users] E1/T1 failover hardware

[John Daragon]

Warning ! I know zip about electronics.

I've been looking for a device to handle the switching of an E1 
connection from one Asterisk box to another in the event of a 
catastrophic server failure.  All of the solutions I've seen so far have 
been designed to handle the situation where the telco line faults so 
that the local PBX can switch to a secondary E1.


I've come across this application note :

http://www.maxim-ic.com/appnotes.cfm/appnote_number/2857

which describes T1/E1/J1, N+1 Redundancy With Analog Switches

These parts are obviously designed to be built into E1 boards - hence, I 
think, the protection circuitry.


Here's the question, then :  what (apart from jumping through regulatory 
hoops) is to stop a simple array of MOSFETS (and a bit of control 
circuitry) implementing a failover switch controlled (say) by a pin on a 
serial or parallel port ?


jd

--

John Daragon  [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127


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Re: [Asterisk-Users] E1 PRI error: !! Got I-frame while link state 2 and !! Got a UA, but i'm in state 1 (long)

[Dinesh Nair]

steve, konstanin,

On 10/20/05 13:56 [EMAIL PROTECTED] said the following:
This boils down to I'm trying to start up the link, but the other side 
seems to think that it IS up.


that's the same conclusion i came to, but why is this happenning ? changing 
loopback cables didnt help either.


clocking, some for external.  If you are using loopback cables, I'd 
suggest setting all the spans for internal (X,0,0,ccs,hdb3[,crc4])


tried two suggestions, one was yours with all spans set for timing=0, and 
the other was to have one set to 0, one set to 1, one set to 2 and one set 
to 3. in both cases, the symptoms persisted.


And the loopback wiring is the pair on 1/2 crossed over to the pair on 
4/5.


aye, double checked this as well.

The part you posted is just where Asterisk is restarting each B-channel.  
More useful would be the part corresponding to the debug messages logged 
above.


there're none for those portions. nothing gets printed, even though pri 
debug span is on. i could turn on pri intense debug, but would anyone be 
able to assist in deciphering it ?


when i try to make a call with the following call file,

Channel: Zap/g4/0193116969
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Context: testplan
Extension: 12
Priority: 1

(the zap channels are set to context=zapin which does,
[zapin]
exten = s,1,Answer()
exten = s,2,Playback(demo-echotest)
exten = s,3,Goto(zapin,s,2)
exten = s,4,Hangup()

[testplan]
exten = 12,1,Answer()
exten = 12,2,Directory(default,localextensions,f)
exten = 12,3,Hangup()

*CLI !cp /tmp/1.call /var/spool/asterisk/outgoing
-- Attempting call on Zap/g4/0193116969 for [EMAIL PROTECTED]:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Going to extension s|1 because of immediate=yes
-- Accepting call from '' to 's' on channel 0/4, span 1
-- Executing Answer(Zap/4-1, ) in new stack
-- Executing Playback(Zap/4-1, demo-echotest) in new stack
-- Playing 'demo-echotest' (language 'en')
Oct 21 06:33:03 NOTICE[280]: channel.c:2166 __ast_request_and_dial: Don't 
know what to do with control frame 15

  == Primary D-Channel on span 1 down
Oct 21 06:33:07 WARNING[280]: chan_zap.c:2265 pri_find_dchan: No D-channels 
available!  Using Primary channel 16 as D-channel anyway!

-- Hungup 'Zap/4-1'
!! Got I-frame while link state 2
  == Primary D-Channel on span 4 up
!! Got I-frame while link state 2
-- Channel 0/4, span 4 got hangup request
  == Primary D-Channel on span 1 up
-- Hungup 'Zap/97-1'

what does Don't know what to do with control frame 15 mean ?

a pri debug on span 4 (a pri_cpe which is looped back to span 1, pri_net) 
shows,


-- Attempting call on Zap/g4/0193116969 for [EMAIL PROTECTED]:1 (Retry 1)
-- Making new call for cr 32771
-- Requested transfer capability: 0x00 - SPEECH
 Protocol Discriminator: Q.931 (8)  len=33
 Call Ref: len= 2 (reference 3/0x3) (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 03 a9 83 99]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 25 ]
 [6c 02 00 c3]
 Calling Number (len= 4) [ Ext: 0  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0)

   Presentation: Number not available (67) '' ]
 [70 0b 80 30 31 39 33 31 31 36 39 36 39]
 Called Number (len=13) [ Ext: 1  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0) '0193116969' ]

 [a1]
 Sending Complete (len= 1)

-- Going to extension s|1 because of immediate=yes
-- Accepting call from '' to 's' on channel 0/25, span 1
-- Executing Answer(Zap/25-1, ) in new stack
-- Executing Playback(Zap/25-1, demo-echotest) in new stack
-- Playing 'demo-echotest' (language 'en')
 Protocol Discriminator: Q.931 (8)  len=10
 Call Ref: len= 2 (reference 3/0x3) (Terminator)
 Message type: CALL PROCEEDING (2)
 [18 03 a9 83 99]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 25 ]
-- Processing IE 24 (cs0, Channel Identification)
Oct 21 06:25:35 NOTICE[123]: channel.c:2166 __ast_request_and_dial: Don't 
know what to do with control frame 15


 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 0/0x0) (Originator)
 Message type: RESTART (70)
 [18 03 a9 83 9a]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   

RE: [Asterisk-Users] E1/T1 failover hardware

[Sergio Serrano]

http://www.junghanns.net/en/ISDNguard_produkt.html


srsergio

-Mensaje original-
De: John Daragon [mailto:[EMAIL PROTECTED] 
Enviado el: jueves, 20 de octubre de 2005 17:24
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] E1/T1 failover hardware

Warning ! I know zip about electronics.

I've been looking for a device to handle the switching of an E1 connection
from one Asterisk box to another in the event of a catastrophic server
failure.  All of the solutions I've seen so far have been designed to handle
the situation where the telco line faults so that the local PBX can switch
to a secondary E1.

I've come across this application note :

http://www.maxim-ic.com/appnotes.cfm/appnote_number/2857

which describes T1/E1/J1, N+1 Redundancy With Analog Switches

These parts are obviously designed to be built into E1 boards - hence, I
think, the protection circuitry.

Here's the question, then :  what (apart from jumping through regulatory
hoops) is to stop a simple array of MOSFETS (and a bit of control
circuitry) implementing a failover switch controlled (say) by a pin on a
serial or parallel port ?

jd

-- 

John Daragon  [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127


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[Asterisk-Users] cdr_odbc with tds

[Ben merrills]

Does anyone know why, using latest cvs head, freetds 0.62.1-0 and
unixODBC, when running cdr_odbc, it says it's logged the call
successfully, however, when checking the table, nothing is there!

I checked through the bug tracker; and a problem very much like mine was
in there, with status resolved as of last year (1339).

Can anyone shed some light on this please?

Cheers,

Ben
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Re: [Asterisk-Users] E1/T1 failover hardware

[Jon Pounder]

 Warning ! I know zip about electronics.

why not just use a multipole relay ?

a 4pole double throw relay gives you 4 sets of contacts for the 2x tx and
2x rx wires. if you want to control with a bit in a parallel port, use
something like a uln2003 relay driver (if the coil current is low enough),
or a couple discrete transistors with the right gain and power handling.

use the 12vdc out of a spare drive connector to power the relay.

I would use one relay rather than 2 dpdt ones so that the switches are
mechanically locked together and if one relay sticks you don't get a weird
combination of circuits connected. Nothing will break, and the phone cops
won't likely bother you if this does happen, but it could be real annoying
and hard to diagnose if it does.

This is basically the electromechanical equivalent of you pulling one
cable and plugging in another (which is what I was going to do with some
T1 routers), except, I found the TXPort.

This actually is meant for failing between telco circuits, but works just
fine working failing between CPE instead. it actually has csus, reframers,
clock generator etc, as well as the relay circuit I describe to do the
switchover. it actually samples the lines and uses some intelligence to
see which to switch to. The device is obsolete so you'll only find it
surplus now, and its t1 only as far as I know but there is probably E1
gear around that does the same thing. I bought mine for $20 so it was not
even worth thinking about my own setup for that price, but they were
listed at up to $3000 when new.





 I've been looking for a device to handle the switching of an E1
 connection from one Asterisk box to another in the event of a
 catastrophic server failure.  All of the solutions I've seen so far have
 been designed to handle the situation where the telco line faults so
 that the local PBX can switch to a secondary E1.

 I've come across this application note :

 http://www.maxim-ic.com/appnotes.cfm/appnote_number/2857

 which describes T1/E1/J1, N+1 Redundancy With Analog Switches

 These parts are obviously designed to be built into E1 boards - hence, I
 think, the protection circuitry.

 Here's the question, then :  what (apart from jumping through regulatory
 hoops) is to stop a simple array of MOSFETS (and a bit of control
 circuitry) implementing a failover switch controlled (say) by a pin on a
 serial or parallel port ?

 jd

 --

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[Asterisk-Users] Re: Why Asterisk documentation is so poor...

[Doug Meredith]

Olle E. Johansson [EMAIL PROTECTED] wrote:

Sergey,
I am sorry if you took our comments that badly. I proposed a worthing
and you did not accept that and refused to update according to our
suggestions. Tilghman therefor decided to close the bug.

I suggest you try again, re-open the bug, fix the problem and continue
to add more documentation. We do need more documentation! It has to be
correct though, and that's why we are giving feedback.

Olle,

I believe I understand and share Sergey's confusion.  Maybe it is
something we just don't understand about how Asterisk development
works.  If he has made a useful contribution with the exception of one
sentence, why don't you just change that sentence and apply it?  Will
you only accept suggestions in the form of directly appliable patches?

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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Re: [Asterisk-Users] Multitenant Call Center Setup

[Waldo Rubinstein]

I have played with AddQueueMember and it works great. However, there is one problem that I have and I hope someone can point me in the right direction.My client's agents rotate seats. This means that if I want to track calls by agent, I can't with AddQueueMember. When I look at the CDR, it tells me the calls made/received by the station (regardless of technology - SIP/AIX/etc). But, at any given point, I don't know which agent made the call.In reality, even with AgentCallBackLogin I can't tell which agent made or received the call. Is there a way that I can identify in the CDR which agent actually received or placed a call regardless of which extension he/she may be sitting on?Thanks,WaldoOn Oct 10, 2005, at 12:22 PM, Waldo Rubinstein wrote:BJ,Thanks for the prompt response. Both my clients work by using the AgentCallBackLogin so that * can send queued calls to them regardless of which SIP phone they're sitting on (sorry I didn't include this in my original email)You mean to say that if I use AddQueueMember, I could do the same and still be able to have two agents 1001?Thanks,WaldoOn Oct 10, 2005, at 11:38 AM, BJ Weschke wrote: There isn't a way to do it in agents.conf.     That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to support a multi-tenant environment.   On 10/10/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Hi list (again),I have another question which I have not been able to resolve fromneither the wiki nor Google. I've been able to configure a multi-tenant setup of asterisk for 2small call centers with no problem, by simply playing with contexts(which I guess is how everyone else is doing it).The problem I have is that I've only been able to configure one global agents.conf file. This restricts my setup in a way that Icannot have two agents 1001, for example if my clients wanted to. Isthere a way to overcome this?Thanks,Waldo___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [Asterisk-Users] E1/T1 failover hardware

[John Daragon]

Sergio Serrano wrote:

http://www.junghanns.net/en/ISDNguard_produkt.html


srsergio

-Mensaje original-
De: John Daragon [mailto:[EMAIL PROTECTED] 
Enviado el: jueves, 20 de octubre de 2005 17:24

Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] E1/T1 failover hardware

Warning ! I know zip about electronics.

I've been looking for a device to handle the switching of an E1 connection
from one Asterisk box to another in the event of a catastrophic server
failure.  All of the solutions I've seen so far have been designed to handle
the situation where the telco line faults so that the local PBX can switch
to a secondary E1.


Thanks Sergio.  I won't need to get the soldering iron out after all.

jd

--

John Daragon  [EMAIL PROTECTED]
argv[0] limited
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Re: [Asterisk-Users] cdr_odbc with tds

[Sergey Okhapkin]

Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a
specialist in ODBC, but what seems to me wrong is the module does INSERT
into the database, but does not make COMMIT.

On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote:
 Does anyone know why, using latest cvs head, freetds 0.62.1-0 and
 unixODBC, when running cdr_odbc, it says it's logged the call
 successfully, however, when checking the table, nothing is there!
 
 I checked through the bug tracker; and a problem very much like mine was
 in there, with status resolved as of last year (1339).
 
 Can anyone shed some light on this please?
 
 Cheers,
 
 Ben
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Re: [Asterisk-Users] SIP to IAX

[Frank Kostin]

yeah yusYu Safin [EMAIL PROTECTED] wrote:
On 10/19/05, Steve Totaro <[EMAIL PROTECTED]>wrote: YES - Original Message - From: "Frank Kostin" <[EMAIL PROTECTED]> To:  Sent: Wednesday, October 19, 2005 8:58 AM Subject: [Asterisk-Users] SIP to IAX Hello everybody, Is it possible to route "any" incoming SIP call (without authentication - register) from an Asterisk A to a remote Asterisk B(throught IAX2), transparently ? Otherwise said, I would like to pass any incoming SIP call from Asterisk A to Asterisk B without SIP need to be registered, like a phone call in zap. I would apreciate any hint, Thanks, Frankshort answer yes,read on,what you really need to know is the compression. You want to
 avoidhaving to compress/uncompress different formats more than once. Inormally have my SIP phones on 711 (same LAN to Asterisk A), then thecalls travel via IAX2 to Asterisk B (yes, it is transparent). FromAsterisk B, they might go to Zap phones, SIP phones, IAX phones, FXO(Zap), channel banks, etc.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System

[Jason Becker]

Vahan Yerkanian wrote:
I'd recommend using native mp3 support that is available in CVS HEAD, as 
madplayer mp3 decoder gives a lower quality sound (audibly more 
cranky/noisy).


I don't follow CVS commits but if that's the case the mpg123 target 
should be removed from the asterisk Makefile and the native mp3 support 
should be documented in ..doc/README.mp3



Jason Becker wrote:


Steve Totaro wrote:


Anyone know how to get around this?  I am stumped.

# make mpg123
[ -f mpg123-0.59r.tar.gz ] || fetch
http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
[ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz
make -C mpg123-0.59r linux



cc1: error: CPU you selected does not support x86-64 instruction set


Use madplayer instead. There are several reasons why Digium  the 
Asterisk community should part ways with mpg123.


Regards,

--
Jason Becker
Director  CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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Re: [Asterisk-Users] E1/T1 failover hardware

[John Daragon]

Jon Pounder wrote:

Warning ! I know zip about electronics.



why not just use a multipole relay ?

a 4pole double throw relay gives you 4 sets of contacts for the 2x tx and
2x rx wires. if you want to control with a bit in a parallel port, use
something like a uln2003 relay driver (if the coil current is low enough),
or a couple discrete transistors with the right gain and power handling.

use the 12vdc out of a spare drive connector to power the relay.

I would use one relay rather than 2 dpdt ones so that the switches are
mechanically locked together and if one relay sticks you don't get a weird
combination of circuits connected. Nothing will break, and the phone cops
won't likely bother you if this does happen, but it could be real annoying
and hard to diagnose if it does.

This is basically the electromechanical equivalent of you pulling one
cable and plugging in another (which is what I was going to do with some
T1 routers), except, I found the TXPort.


Good idea.  I just have an irrational dislike of moving parts. And I 
*like* MOSFETS !



This actually is meant for failing between telco circuits, but works just
fine working failing between CPE instead. it actually has csus, reframers,
clock generator etc, as well as the relay circuit I describe to do the
switchover. it actually samples the lines and uses some intelligence to
see which to switch to. The device is obsolete so you'll only find it
surplus now, and its t1 only as far as I know but there is probably E1
gear around that does the same thing. I bought mine for $20 so it was not
even worth thinking about my own setup for that price, but they were
listed at up to $3000 when new.


I've only had a quick look for these, but E1 ones seem to be thin on the 
ground and expensive, and I have a horrible feeling that all the 
reframing stuff just adds another set of variables if something goes 
wrong somewhere.


Thanks again.

jd

--

John Daragon  [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
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RE: [Asterisk-Users] Help with Dial Plan

[Dave Morrow]

Thanks Steve, the 'w's worked great. I managed to tune it down to them
only hearing a please wait out of the greeting.. 


David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615 

 Poor planning on your part does not necessarily constitute an
emergency on my part! 

This message has originated from Autodata Solutions. The attached
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 19, 2005 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with Dial Plan



On Wed, 19 Oct 2005, Dave Morrow wrote:

 Thanks Steve.  It almost works, but never dials the extension.  Also, 
 is there a way I could mute the line while the remote attendant comes
on?


Oops sorry - the dangers of posting without testing.

The ,s are wrong - they should be w.  Each w is 1/2 second of waiting.

So that makes it:

exten = _6XXX,1,Dial(Zap/gX/1234567890,60,D(${EXTEN}))

As for the muting - bit of a loss about that one.

Steve

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RE: [Asterisk-Users] cdr_odbc with tds

[Ben merrills]

What should I do? :)

Add it to the bug tracker?

Ben

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey
Okhapkin
Sent: 20 October 2005 16:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] cdr_odbc with tds

Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a
specialist in ODBC, but what seems to me wrong is the module does INSERT
into the database, but does not make COMMIT.

On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote:
 Does anyone know why, using latest cvs head, freetds 0.62.1-0 and
 unixODBC, when running cdr_odbc, it says it's logged the call
 successfully, however, when checking the table, nothing is there!
 
 I checked through the bug tracker; and a problem very much like mine
was
 in there, with status resolved as of last year (1339).
 
 Can anyone shed some light on this please?
 
 Cheers,
 
 Ben
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Re: [Asterisk-Users] Re: Why Asterisk documentation is so poor...

[Olle E. Johansson]

Doug Meredith wrote:
 Olle E. Johansson [EMAIL PROTECTED] wrote:
 
 
Sergey,
I am sorry if you took our comments that badly. I proposed a worthing
and you did not accept that and refused to update according to our
suggestions. Tilghman therefor decided to close the bug.

I suggest you try again, re-open the bug, fix the problem and continue
to add more documentation. We do need more documentation! It has to be
correct though, and that's why we are giving feedback.
 
 
 I believe I understand and share Sergey's confusion.  Maybe it is
 something we just don't understand about how Asterisk development
 works.  If he has made a useful contribution with the exception of one
 sentence, why don't you just change that sentence and apply it?  Will
 you only accept suggestions in the form of directly appliable patches?
 
Well, as Corydon76 said in the bug report - neither he or I can commit
patches. Neither of us are paid by anyone to spend our time fixing other
people's patches or bugs or even giving input... So please don't suppose
that we have time or require us to fix other peoples additions.

And with the pressure on the committers and developers that we have now,
while trying to stabilize 1.2, we do rely on the reporter to try to take
the patch all the way, considering input. It was an easy change to make,
and should not have caused this discussion.

I do not necessarily agree that the bug should have been closed, I would
personally have kept it open until someone bothered with doing the
necessary changes and moved it forward.

We do need more people that are interested in writing docs and fixing
bugs, not just developers that add new features. Very few spend time
fixing other people's patches, testing other people's patches, giving
input and assisting in moving stuff forward. Very few people outside
Digium fix bugs reported in the bug tracker, many more contribute new
code for new functionality.

Feel free to join the larger development group, helping all of us to
move forward. Try to balance being a user that gets something for free
with contributing back :-)

If you want to fix this particular document, just find someone in the
#asterisk-bugs channel on IRC and we'll happily re-open the bug report
for you.

/Olle

...who has had to fix many documentation contributions several times
before they where accepted...
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Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System

[Olle E. Johansson]

Jason Becker wrote:
 Vahan Yerkanian wrote:
 
 I'd recommend using native mp3 support that is available in CVS HEAD,
 as madplayer mp3 decoder gives a lower quality sound (audibly more
 cranky/noisy).
 
 
 I don't follow CVS commits but if that's the case the mpg123 target
 should be removed from the asterisk Makefile and the native mp3 support
 should be documented in ..doc/README.mp3
 
Go ahead and submit a patch to the README to the bug tracker. I will try
to keep Corydon from closing it :-)

/O
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[Asterisk-Users] D-Link DG104S firmware upgrade for flash funcionality on *

[Andrea Frigo]

Hy guys,
 I'm trying to upgrade the 
firmware of this gateway to get the flash digit work. Whit my version of 
firmware the flash signal is interpreted on Asterisk as the number 1, so I'm 
looking for a firmware upgrade to solve the problem. I read on this list that 
some of you had the same problem and solve it upgrading form the Fw 3.0B35 to 
the 3.0B44, but the link they refer to is now broken.
May besome one have the file and can mail it 
to me or point me to the right link.
I need to upgrade two types of DG104S 
gatways:
 Hardware revision 
C1
 Boot 3.0B14-C
 Firmware 3.0B35-C
and
 Hardware revision 
D1
 Boot 4.0-B09
 Firmware 4.0-B28
all with the same problem.

Regards,
 Andrea 
Frigo
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RE: [Asterisk-Users] cdr_odbc with tds

[Jon Pounder]

 What should I do? :)

 Add it to the bug tracker?

it might be a bug, but I don't think its due to lack of commit.

sqlserver is normally in implicit commit mode where every sql statement
is an individual transaction and is committed as its executed.

I would start by having a look at the driver and database logs and see
what is actually being executed first, then go from there.



 Ben

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Sergey
 Okhapkin
 Sent: 20 October 2005 16:48
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] cdr_odbc with tds

 Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a
 specialist in ODBC, but what seems to me wrong is the module does INSERT
 into the database, but does not make COMMIT.

 On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote:
 Does anyone know why, using latest cvs head, freetds 0.62.1-0 and
 unixODBC, when running cdr_odbc, it says it's logged the call
 successfully, however, when checking the table, nothing is there!

 I checked through the bug tracker; and a problem very much like mine
 was
 in there, with status resolved as of last year (1339).

 Can anyone shed some light on this please?

 Cheers,

 Ben
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Jon Pounder

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RE: [Asterisk-Users] Re: Why Asterisk documentation is so poor...

[Richard Cook]

A great stance.  Another contributor most likely lost.  Nice job.

--
R


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Thursday, October 20, 2005 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Why Asterisk documentation is so poor...

Doug Meredith wrote:
 Olle E. Johansson [EMAIL PROTECTED] wrote:
 
 
Sergey,
I am sorry if you took our comments that badly. I proposed a worthing 
and you did not accept that and refused to update according to our 
suggestions. Tilghman therefor decided to close the bug.

I suggest you try again, re-open the bug, fix the problem and continue 
to add more documentation. We do need more documentation! It has to be 
correct though, and that's why we are giving feedback.
 
 
 I believe I understand and share Sergey's confusion.  Maybe it is 
 something we just don't understand about how Asterisk development 
 works.  If he has made a useful contribution with the exception of one 
 sentence, why don't you just change that sentence and apply it?  Will 
 you only accept suggestions in the form of directly appliable patches?
 
Well, as Corydon76 said in the bug report - neither he or I can commit
patches. Neither of us are paid by anyone to spend our time fixing other
people's patches or bugs or even giving input... So please don't suppose
that we have time or require us to fix other peoples additions.

And with the pressure on the committers and developers that we have now,
while trying to stabilize 1.2, we do rely on the reporter to try to take the
patch all the way, considering input. It was an easy change to make, and
should not have caused this discussion.

I do not necessarily agree that the bug should have been closed, I would
personally have kept it open until someone bothered with doing the necessary
changes and moved it forward.

We do need more people that are interested in writing docs and fixing bugs,
not just developers that add new features. Very few spend time fixing other
people's patches, testing other people's patches, giving input and assisting
in moving stuff forward. Very few people outside Digium fix bugs reported in
the bug tracker, many more contribute new code for new functionality.

Feel free to join the larger development group, helping all of us to move
forward. Try to balance being a user that gets something for free with
contributing back :-)

If you want to fix this particular document, just find someone in the
#asterisk-bugs channel on IRC and we'll happily re-open the bug report for
you.

/Olle

...who has had to fix many documentation contributions several times before
they where accepted...
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RE: [Asterisk-Users] cdr_odbc with tds

[Sergey Okhapkin]

...Or fix the problem yourself:-)

On Thu, 2005-10-20 at 16:58 +0100, Ben merrills wrote:
 What should I do? :)
 
 Add it to the bug tracker?
 
 Ben
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Sergey
 Okhapkin
 Sent: 20 October 2005 16:48
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] cdr_odbc with tds
 
 Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a
 specialist in ODBC, but what seems to me wrong is the module does INSERT
 into the database, but does not make COMMIT.
 
 On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote:
  Does anyone know why, using latest cvs head, freetds 0.62.1-0 and
  unixODBC, when running cdr_odbc, it says it's logged the call
  successfully, however, when checking the table, nothing is there!
  
  I checked through the bug tracker; and a problem very much like mine
 was
  in there, with status resolved as of last year (1339).
  
  Can anyone shed some light on this please?
  
  Cheers,
  
  Ben
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[Asterisk-Users] Asterisk Community Participant; Katrina Refugee UPDATE

[JR Richardson]

Hi all,



Thank you all for your replies of hope, and advice for
recovering flooded computer equipment. I was not able to recover ANY
electronic components. There was 5 foot of water sitting in my home for
over a week. The water was laden with very corrosive contaminants and heavy
sludge. Literally this water ate the conformal coating off a lot of ckt
boards then heavily corroded and oxidized any metal and solder. I
struggled to recover data from hard drives but did manage to get a couple to
work after cleaning.



Silver Lining:



I lost a few thousand dollars just in Digium hardware alone,
so I contacted Digium and let them know that I was out of commission for a
while till I could get another lab setup. In my most humble manner I requested
card replacement or discount on a few items just so I could get back up and
running, to my surprise, Digium would not accept any money from me and sent
several cards and components free. They thanked me for my participation
and expressed compassion for my loss. Digium is a class act and have helped
me rebuild some of what Katrina knocked down. I will forever be grateful and
never forget Digiums Support in my time of need.



Thank you; Mark and Malcolm, you guys are the best.



I have already relocated my family to Lafayette, LA and we are
living in the home we intend to buy soon, just need to run some extra power
ckts to the spare room and setup a new lab. We took in a family that
lived across the street from us in St Bernard and they are living in my future
computer lab, so it will still be a bit before I get up and running but will
soon.



JR Richardson






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Re: [Asterisk-Users] cdr_odbc with tds

[Andy Kuo]

What database server are you using?
If you are using MSSQL, just use freetds without unixODBC.

AK
On 10/20/05, Ben merrills [EMAIL PROTECTED] wrote:
Does anyone know why, using latest cvs head, freetds 0.62.1-0 andunixODBC, when running cdr_odbc, it says it's logged the call
successfully, however, when checking the table, nothing is there!I checked through the bug tracker; and a problem very much like mine wasin there, with status resolved as of last year (1339).Can anyone shed some light on this please?
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[Asterisk-Users] Re: Why Asterisk documentation is so poor...

[BJ Weschke]

 Richard,

 I'm sorry you and others feel the way you do. Businesses though don't
want an open source project that is a free for all when it comes to
contributions and discipline both in the code itself and
documentation.

 Olle has contributed hundreds of hours of his own time over the time
he's been involved with Asterisk. It's a disappointment that you've
taken to mocking his efforts instead of taking the suggestion and
making a contribution that works for everyone. We all agree readily
that the existing documentation could use the enhancement.


On 10/20/05, Richard Cook [EMAIL PROTECTED] wrote:

 A great stance.  Another contributor most likely lost.  Nice job.

 --
 R


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
 Johansson
 Sent: Thursday, October 20, 2005 12:01 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Re: Why Asterisk documentation is so poor...

 Doug Meredith wrote:
  Olle E. Johansson [EMAIL PROTECTED] wrote:
 
 
 Sergey,
 I am sorry if you took our comments that badly. I proposed a worthing
 and you did not accept that and refused to update according to our
 suggestions. Tilghman therefor decided to close the bug.
 
 I suggest you try again, re-open the bug, fix the problem and continue
 to add more documentation. We do need more documentation! It has to be
 correct though, and that's why we are giving feedback.
 
 
  I believe I understand and share Sergey's confusion.  Maybe it is
  something we just don't understand about how Asterisk development
  works.  If he has made a useful contribution with the exception of one
  sentence, why don't you just change that sentence and apply it?  Will
  you only accept suggestions in the form of directly appliable patches?
 
 Well, as Corydon76 said in the bug report - neither he or I can commit
 patches. Neither of us are paid by anyone to spend our time fixing other
 people's patches or bugs or even giving input... So please don't suppose
 that we have time or require us to fix other peoples additions.

 And with the pressure on the committers and developers that we have now,
 while trying to stabilize 1.2, we do rely on the reporter to try to take
 the
 patch all the way, considering input. It was an easy change to make, and
 should not have caused this discussion.

 I do not necessarily agree that the bug should have been closed, I would
 personally have kept it open until someone bothered with doing the
 necessary
 changes and moved it forward.

 We do need more people that are interested in writing docs and fixing bugs,
 not just developers that add new features. Very few spend time fixing other
 people's patches, testing other people's patches, giving input and
 assisting
 in moving stuff forward. Very few people outside Digium fix bugs reported
 in
 the bug tracker, many more contribute new code for new functionality.

 Feel free to join the larger development group, helping all of us to move
 forward. Try to balance being a user that gets something for free with
 contributing back :-)

 If you want to fix this particular document, just find someone in the
 #asterisk-bugs channel on IRC and we'll happily re-open the bug report for
 you.

 /Olle

 ...who has had to fix many documentation contributions several times before
 they where accepted...
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[Asterisk-Users] Some questions regarding T1's

[Michaël Gaudette]

Hi,

I'm a computer engineer with basic knowledge of telecom.  Actually, less 
then basic to be honest.  I've been playing around with Asterisks for a few 
weeks with 2 FXS and 2 FXO cards, and having a bit of fun making a home PBX. 
I'd like to know how I could apply this new knowledge to, for example, 
developping a PBX solution for this following hypothetical company:


- Exactly 72 employees each with a direct telephone number that goes 
directly to their phone.  Ex: Bob is 444-555- and Lisa is 444-555-6667. 
Let's say they don't have a PBX yet.
- Statistically, the max number of outside lines ever busy at the same time 
was 24 (how conveniently T1-like).  They don't want to change their business 
cards, so 444-555- should still reach Bob, but now by going to the PBX 
first.  The PBX should recognize that the call was made to 444-555- and 
switch it to Bob automatically.  Bob should see the Caller ID of the caller 
on his phone.


This is it.  Conceptually, not very complicated.  My guess is I would need 
(and this is where I need confirmation from somebody in the know):

- Asterisk PBX
- A Digium T1 line for a connection to the phone service provider (I'm in 
Canada, so let's say Bell Canada for argument's sake)

- A T1 line from Bell Canada (or other)
- Something (not sure what) on the outside to connect to those 72 phones (3 
T1 cards internally connecting to a wire panel, in turn connected to 60 
phones?


Is this it?  Do I need anything else?

Follow-up questions:
a) Is is possible to have 72 numbers associated to a single T1 (more numbers 
than lines)?
b) Will Asterisk be able to recognize (and how?) which number the call came 
on, so it can run the right dial plan?
c) This migth be a Canada-specific answer, but I'll try:  When leasing a T1 
line, does the regional code  have to be based on geohraphy?  Could I have a 
T1 with 416 (Toronto) numbers located in Montreal (514)?


I sure hope my questions weren't too newbie-like.  I fear they are, but 
I've really tried finding the info on  the web. I certainly wouldn't be 
insulted if the only reply I got was a link to a decent Web site explaining 
all this.


Mike 


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