date:20060915

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Douglas Garstang wrote:
 The docs at that URL say that the dictionary has 'yes' in it... although I 
 don't understand how I can get replies like 'YOU HALF' if it doesn't exist in 
 the dictionary.

Did you read the Sphinx documentation?

Rather heavy but describe everything!

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] Bri Card for Asterisk ?

2006-09-15 Thread stoffell


On 9/15/06, Noc Phibee [EMAIL PROTECTED] wrote:

a small question:
what is the best card for Asterisk for supply 2/4 BRI access to a old PABX ?


A good bri card is the quadbri of Junghanns/Beronet or Digium (haven't
tried the Digium one, but seems interesting because of the on-board
echo can..).

cheers
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Re: [asterisk-users] Re: PRI: sometimes Asterisk drop calls


Hi,
I do not use queues but I have a lot of messages like that. I talked a 
lot with Steve about this
It seems like Asterisk cannot agree with telco about which channels are 
busy and which are not. Maybe a bug? I do not know...it seems too 
strange Asterisk has a so big problem. There must be something we do 
not knowBy the way, the solution seems to be using the higher 
channels of the span, in other words to make calls using G instead of g 
inside Dial command (thans to Steve and others!!)


Hope may help!

Giorgio Incantalupo



Tomislav Parčina wrote:

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
  

Do you have queues/agents configured?



No, I don't have queues nor agents configured.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
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Re: [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?

2006-09-15 Thread Christian Mohrbacher

In some cases : Yes.
But we have the following situation : We re using cisco 7960 phones in
each office (about 150 of them), but not every person has it's own
phone. Normally there are two employees in one office and they share one
phone, BUT have their own extension. Fortunately the Ciscos are able to
register each line seperately. The lines are registered via their
extension, which means if you configure your phone to have extension
1234 the line will be registeres as SIP/1234.
Now we want to give the user's the ability to take their number with
them. So when you change places you can call a defined number which
will write you a config file for your new phone.
Now, if I have extension 1234 and go to a different office, or to a
meeting room, etc and log into that phone using my extension, if i did
not log out my normal phone we have a problem because we have to SIP/1234.
I haven't found a good solution for that yet, but if I could register
two SIP/1234 phones the problem would be solved.

Brandon Galbraith wrote:
 Can't this be solved with one extension simply ringing two different
 SIP devices?

 -brandon

 On 9/15/06, *Christian Mohrbacher* 
 [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 Hi Oliver,
 just one advantage of multiple registrations : Imagine you are
 working
 in two different departments with your time split 50/50. Now you
 have to
 different offices. You have an office in department A but when working
 for department B you are at a different one.
 Now you want your personal phone number ringing in both offices at
 the
 same time. This would be much easier if asterisk learns how to
 deal with
 multiple registrations.


 Olivier wrote:
  Hi,
 
  I've read many times  multiple registration is an important missing
  feature in Asterisk.
  I'm not sure I've understood the reason(s) behind that.
 
  Could you explain ?
 
  Cheers
 
 

 
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 -- 
 Brandon Galbraith
 Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 AIM: brandong00
 Voice: 630.400.6992
 A true pirate starts drinking before the sun hits the yard-arm.
 Ya. --thelost
 

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Re: [asterisk-users] How to download asterisk 1.3 development version

-BEGIN PGP SIGNED MESSAGE-
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Boneym wrote:
 Hi,
 
  I would like to test asterisk 1.4 development version , can anyone send me
 a link to it . Thanks in advance.

This would be SVN trunk (http://www.asterisk.org/download):

Commands to check out code from our SVN repository:

# cd /usr/src

# svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
# svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
# svn checkout http://svn.digium.com/svn/libpri/trunk libpri

- --
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Matt Riddell
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Re: [asterisk-users] Too many files... error - best way to fix?

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Matt Arnilo S. Baluyos (Mailing Lists) wrote:
 On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
 Matt Arnilo S. Baluyos (Mailing Lists) wrote:
  Hello everyone,
  What would be the best way to solve this error on ARI?
  We are using ARI version 00.08.04 on an [EMAIL PROTECTED] server.
 
 Check the asterisk readme.
 
 Hello Steve,
 
 The closest thing regarding this error on the README is the one on
 increasing the nofiles on the /etc/security/limits.conf file and
 then rebooting.

Type

ulimit -n 8192

before starting Asterisk.

ulimit -n without a number will tell you what it is currently set to.

- --
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Matt Riddell
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[asterisk-users] non-technical, dealing with users giving feedback

2006-09-15 Thread stoffell


Hi list,

Any suggestions on how to deal correctly (socially and technically)
with users complaining about features/issues? For instance, users
complaining about echo; personally I ask the user(s) to give me all
the details when reporting echo (like; using handset/speaker,
internal/external call, volume, location, time, phone number, ..).

This goes well the very first time, but the users (and I understand
that) forget some details, or after a few times, don't give the
details anymore.. Result, troubleshooting gets much harder.

Is there a good way to deal with these situations without annoying the
users or myself too much?

Thanks for the tips..
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[asterisk-users] two safe_asterisk processes on the same PBX???


Hi,
I noticed sometimes I get the messages remote unix connection every 
1or 2 seconds. I found that there is a second safe-asterisk process 
which is probably trying to start/connect to asterisk.

Is there anybody who knows why (and maybe how to solve it)?

TIA


Giorgio Incantalupo
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[asterisk-users] Re: PRI: sometimes Asterisk drop calls

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Hi,
 I do not use queues but I have a lot of messages like that. I talked a 
 lot with Steve about this
 It seems like Asterisk cannot agree with telco about which channels are 
 busy and which are not. Maybe a bug? I do not know...it seems too 
 strange Asterisk has a so big problem. There must be something we do 
 not knowBy the way, the solution seems to be using the higher 
 channels of the span, in other words to make calls using G instead of g 
 inside Dial command (thans to Steve and others!!)

I don't think that could be the problem. Because Asterisk has already 
established connection with provider on certain channel. So why would they 
negotiate another channel? When I transfer phone call to another extension, 
incoming channel doesn't change.

I think something else is the problem, but I do encourage to use G in 
dialplan's Dial command.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
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Re: [asterisk-users] Correct settings for UK (BT) FXO

2006-09-15 Thread Gordon Henderson

On Thu, 14 Sep 2006, Faris Raouf wrote:

 Incidentally I think there are people on this list who have no issues
 with the TDM400p in the UK, but I have no idea how/why.

I have a small number of TDM400P's in the field - all with 1 FXO and 1 FXS
port, and it seems to just work, although once or twice it does failt to
detect a hangup or the remote end, and the voicemail records some
 at the end of the message - but my old answering machine
used to do that too!

The only issue I have is routing calls through the TDM card from the
incoming to the outgoing ports, when I have fax detection turned on, the
internal analogue phones are overly loud and seem to suffer some sort of
local echo/reverb problem.

Configs below:

Gordon


Zaptel.conf:

fxsks=1
fxoks=4
loadzone=uk
defaultzone=uk


zapata.conf:

[trunkgroups]

[channels]
usecallerid=yes
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity ; Added for UK CLI detection
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
;faxdetect=incoming

context=internal
signalling=fxo_ks
sendcalleridafter=2 ; the hidden option given by Slav from Digium
mailbox=100
callerid=100
channel = 4

context=incoming
signalling=fxs_ks
;rxgain=7.0 ; Need this for incoming FAXes
;txgain=7.0
immediate=no ; will take the line on the first ring
callerid=asreceived ; propagate the CID received from BT
channel = 1
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Re: [asterisk-users] non-technical, dealing with users giving feedback

-BEGIN PGP SIGNED MESSAGE-
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stoffell wrote:
 Hi list,
 
 Any suggestions on how to deal correctly (socially and technically)
 with users complaining about features/issues? For instance, users
 complaining about echo; personally I ask the user(s) to give me all
 the details when reporting echo (like; using handset/speaker,
 internal/external call, volume, location, time, phone number, ..).
 
 This goes well the very first time, but the users (and I understand
 that) forget some details, or after a few times, don't give the
 details anymore.. Result, troubleshooting gets much harder.
 
 Is there a good way to deal with these situations without annoying the
 users or myself too much?

Search Daily Asterisk News for echo:

http://www.sineapps.com/news.php?rssid=1437

:)

- --
Cheers,

Matt Riddell
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[asterisk-users] Modem calls

2006-09-15 Thread Jose Limeres

Hi,
I need to pass modem calls through a TDM400 card. Conecting the modem
to the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4)
directly.
Even though Echo cancellation is disabled in both lines the call is
never successful. Modems speak for some time and then the line is hang
up with the message No answer from remote side. I send thereafter
some traces in case someone is interested.

I know modem calls are not easy as something is messed up in the
signaling going through asterisk but has someone been successful in an
easy configuration such as this one?

Thanks,
Jose


Sep 15 09:45:22 DEBUG[4625] chan_zap.c: Dialing '908274101'

Sep 15 09:45:22 DEBUG[4625] chan_zap.c: Deferring dialing...

Sep 15 09:45:22 DEBUG[4625] chan_zap.c: Requested indication 3 on channel Zap/1-1

Sep 15 09:45:23 DEBUG[4625] chan_zap.c: Exception on 15, channel 4

Sep 15 09:45:23 DEBUG[4625] chan_zap.c: Got event Hook Transition Complete(12) on channel 4 (index 0)

Sep 15 09:45:25 DEBUG[4625] chan_zap.c: Exception on 15, channel 4

Sep 15 09:45:25 DEBUG[4625] chan_zap.c: Got event Dial Complete(9) on channel 4 (index 0)

Sep 15 09:45:25 DEBUG[4625] chan_zap.c: No echo cancellation requested

Sep 15 09:45:25 DEBUG[4625] chan_zap.c: No echo training requested

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Exception on 15, channel 4

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Got event Dial Complete(9) on channel 4 (index 0)

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: No echo cancellation requested

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Requested indication -1 on channel Zap/1-1

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Took Zap/1-1 off hook

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: master: 1, slave: 4, nothingok: 0

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Stopping tones on 1/0 talking to 4/0

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Stopping tones on 4/0 talking to 1/0

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Making 4 slave to master 1 at 0

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Added 15 to conference 9/1

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Added 13 to conference 9/4

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Updated conferencing on 1, with 0 conference users

Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Updated conferencing on 4, with 0 conference users

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Exception on 13, channel 1

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Got event On hook(1) on channel 1 (index 0)

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: No echo cancellation requested

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: No echo cancellation requested

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Unlinking slave 4 from 1

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Removed 15 from conference 9/1

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Removed 13 from conference 9/4

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Updated conferencing on 1, with 0 conference users

Sep 15 09:45:51 DEBUG[4625] channel.c: Returning from native bridge, channels: Zap/1-1, Zap/4-1

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Hangup: channel: 4 index = 0, normal = 15, callwait = -1, thirdcall = -1

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/4-1

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Updated conferencing on 4, with 0 conference users

Sep 15 09:45:51 DEBUG[4625] app_dial.c: Exiting with DIALSTATUS=ANSWER.

Sep 15 09:45:51 DEBUG[4625] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.

Sep 15 09:45:51 DEBUG[4625] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-09-15 09:45:22','\FXS\
505','505','0908274101','from-internal',
'Zap/1-1','Zap/4-1','Dial','ZAP/4/908274101|120|r',29,24,'ANSWERED',3,'','asterisk-3279-1158306315.0')

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Hangup: channel: 1 index = 0, normal = 13, callwait = -1, thirdcall = -1

Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1


asterisk1*CLI zap show channel 1
Channel: 1CLI
File Descriptor: 13
Span: 1
Extension:
Dialing: no
Context: from-internal
Caller ID: 505
Calling TON: 0
Caller ID name: FXS
Destroy: 0
InAlarm: 0
Signalling Type: FXO Kewlstart
Radio: 0
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 0 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Onhook
asterisk1*CLI

asterisk1*CLI
asterisk1*CLI
asterisk1*CLI zap show channel 4
Channel: 4CLI
File Descriptor: 15
Span: 1
Extension:
Dialing: no
Context: from-zaptel
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP:

[asterisk-users] Re: Maximum retries exceeded on transmission

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 I have searched this list and others, and see other pepole having this
 issue. However, I have not seen how to fix it.
 
 Sep 12 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum
 retries exceeded on transmission
 [EMAIL PROTECTED] for seqno 1620 (Critical
 Response)
 
 Sep 12 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up
 call [EMAIL PROTECTED] no reply to our critical
 packet.
 
 What is the critical packet that is not being responded to? Please help.

I head this problem with SJ phone softphone on one installation. I have 
uninstalled soft phone's and now I use hard phones (Grandsteram GXP 2000) and I 
don't get that errors anymore.

Hope this helps. If you find what exactly is the problem, please let me know.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
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[asterisk-users] Re: BLF across asterisk trunks

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 I second this wish.

I third this wish :))


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
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[Asterisk-Users] Shared Line Appearance, Snom and trunk

2006-09-15 Thread Olivier

Hi,Who has ever programmed Shared Line Appearance option which is due with Asterisk 1.4 ?This feature should be in the trunk but I didn't dare to try it.Is it foreseeable to use it with Snom phones as this phones support SLA ?
Regards
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Re: [asterisk-users] voicemail access thru apache on another server

2006-09-15 Thread Benjamin Jacob


okk.. got it working.
the problem was that I had started out with Realtime, using Mysql.
Seems u can't use mysql and then put in odbc solely for voicemail 
storage. res_odbc.conf entry decides that u r gonna use odbc for everything.

so had to replace mysql stuff with odbc in the conf files.

Now, how on earth do i read the recordings and play them out  thru a 
browser!!


pheww. it never ends!!!


Benjamin Jacob wrote:


THanks RR,
Am trying it right now.
But getting into all kinds of trouble!! ranging from SQL Alloc failed 
- to seg faults.
I honestly donno anything abt odbc(n i seriously dont have the time to 
rnd on that one, right now)


Can you or anyone else paste their config files for(related to 
voicemail odbc storage)

voicemail.conf
odbc.ini
odbcinst.ini
res_odbc.conf
res_mysql.conf (i dont think this should change , was using realtime 
for storing voicemail users and sip users,etc, perfectly)

how abt extconfig.conf (i guess this too duznt change).

all help really appreciated

Ben.

RR wrote:


have a look at Wiki for asterisk +  odbc storage. The database for
storing entire voicemail messages can be stored on a local or a remote
database. Then you can do whatever you want with it. You will have to
recompile asterisk by turning on ODBC storage. It's all there on the
Wiki
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Re: [asterisk-users] 9 becomes 99 ? And other strangeness

On Thu, Sep 14, 2006 at 10:37:59AM -0500, Rich Adamson wrote:
 Try the above an see what the result is. If it does not address the 
 problem, at least one item has been removed from the list of 
 possibilities. ;)

OK, I can now replicate this without using outbound dialing at all, with a
tiny dialplan (attached). I've turned on dtmf logging in logger.conf, and
it's definitely a DTMF problem.

What happens if I dial 611: around 30% of the time it says I'm sorry,
that is not a valid extension, thinking I've dialled 6611. The rest of the
time it works, and I get Hello, world!

Here's what I see on the console when it fails:

   -- Starting simple switch on 'Zap/1-1'
[Sep 15 09:48:00] DTMF[5744]: channel.c:2065 __ast_read: DTMF end '6' received 
on Zap/1-1
[Sep 15 09:48:00] DTMF[5744]: channel.c:2065 __ast_read: DTMF end '6' received 
on Zap/1-1
[Sep 15 09:48:01] DTMF[5744]: channel.c:2065 __ast_read: DTMF end '1' received 
on Zap/1-1
[Sep 15 09:48:01] DTMF[5744]: channel.c:2065 __ast_read: DTMF end '1' received 
on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack
-- Executing [EMAIL PROTECTED]:2] Playback(Zap/1-1, pbx-invalid) in new 
stack
-- Playing 'pbx-invalid' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, s|1) in new stack
-- Goto (internal,s,1)
[Sep 15 09:48:08] WARNING[5744]: pbx.c:2322 __ast_pbx_run: Channel 'Zap/1-1' 
sent into invalid extension 's' in context 'internal', but no invalid handler
-- Hungup 'Zap/1-1'

So my single 6 is read as 66.

Searching the web, the only tweak I can see is to set relaxdtmf=yes in
zapata.conf for each channel. I've now tried that, and if anything, it made
it worse: I got a failure rate of about 50%, and in one case 66611 instead
of 611.

So I'm a bit stuck as to what to do now. My /etc/asterisk/zapata.conf and
/etc/zaptel.conf are also attached for reference.

The only things I can think of:

(1) line level needs tweaking? Is there a way to measure the incoming level
when I dial DTMF, to see if it's too low or is clipping?

(2) I get the following error when first loading the wctdm driver, and I
don't know if it's a problem

-
[EMAIL PROTECTED] asterisk]# modprobe zaptel
[EMAIL PROTECTED] asterisk]# modprobe wctdm
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for wctdm
-

However it seems to work on second attempt:

-
[EMAIL PROTECTED] asterisk]# modprobe wctdm
[EMAIL PROTECTED] asterisk]# ztcfg -vv

Zaptel Version: SVN-trunk-r1459
Echo Canceller: MG2
Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

3 channels configured.

Changing signalling on channel 1 from Unused to FXO Kewlstart
Changing signalling on channel 2 from Unused to FXO Kewlstart
Changing signalling on channel 4 from Unused to FXS Kewlstart
-

Looking at postings on the Asterisk list, it seems that under CentOS I
should just run '/etc/rc.d/init.d/asterisk start' and it will load the
modules automatically - but that doesn't work for me.

That is: if I stop asterisk, rmmod the manuals by hand, and then do:

[EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk start
Starting asterisk: [  OK  ]
[EMAIL PROTECTED] asterisk]# lsmod | grep -i zap
[EMAIL PROTECTED] asterisk]# lsmod | grep -i wct

you can see that they've not been loaded.

My /etc/asterisk/modules.conf is the stock standard one and has autoload=yes
(although I guess that refers to asterisk modules, not kernel modules)

Is there any more debugging I can turn on to find out what might be
happening?

Regards,

Brian.
[internal]

exten = 611,1,Answer()
exten = 611,2,Playback(hello-world)
exten = 611,3,Hangup()
exten = _X.,1,Answer()
exten = _X.,2,Playback(pbx-invalid)
exten = _X.,3,Goto(s,1)
[trunkgroups]
; define any trunk groups

[channels]
;default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
answeronpolarityswitch=no

; define channels
context=internal
signalling=fxo_ks
channel = 1

context=internal
signalling=fxo_ks
channel = 2

context=incoming
signalling=fxs_ks   ; Use FXS signalling for an FXO channel
channel = 4

# Ports 1 and 2 are FXS (FXO signalling)
fxoks=1-2
# Port 4 is FXO (FXS signalling)
fxsks=4

loadzone=uk
defaultzone=uk
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[asterisk-users] RE: Asterisk 1.4 Docs

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 No mention of Shared Line Appearance in the v1.4 new release.  Anyone know
 if they still plan to include it or not?  Digium has been kind of quiet on
 their work on that feature.
 
 With their new Asterisk appliance running v1.4 I certainly hope they have
 SLA as all other traditional/proprietary PBX's in that market segment do.

Yes, and I'm interested in AOC messages. If I'm only able to manipulate with 
them, store them somewhere.

I believe every Asterisk user will benefit with this, it just that people are 
not familiar what AOC does. AOC messages (Advice of charge) are messages that 
your provider sends you at the end of call. They tell you how much units jour 
provider will charge you for that call. And if you would like to know how much 
money is that, you simply multiply it with price of every unit.

It will solve charging problems with Asterisk! We wouldn't have to keep up to 
date our databases with prices. Provider will directly tell us how much he will 
charge every our call.

How to help/motivate developers to work on AOC in Asterisk?


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] AOC - advice of charge

I'm interested in AOC (Advice of Charge) messages in Asterisk.

As far as I know, * does get AOC messages, but it's unable to do anything with 
them. What I would like to know is:
- what is current status of AOC in Asterisk?
- is there any work going on AOC in Asterisk?
- is there anything I could do to make thing go faster in developing AOC in 
Asterisk? (unfortunately I'm not programmer)

What I would like to be able to do with AOC messages is to manipulate with them 
and to store them in CDR or in some other database so that I could do billing. 

I believe every Asterisk user will benefit with this, it just that people are 
not familiar what AOC does. AOC messages (Advice of charge) are messages that 
your provider sends you during or at the end of call. Provider can send you 
charging Info in currency or charging Info pulse.

Come on guy, lets make our life's a little bit easier!


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] two safe_asterisk processes on the same PBX???

2006-09-15 Thread Tzafrir Cohen

On Fri, Sep 15, 2006 at 09:52:02AM +0200, Giorgio Incantalupo wrote:
 Hi,
 I noticed sometimes I get the messages remote unix connection every 
 1or 2 seconds. I found that there is a second safe-asterisk process 
 which is probably trying to start/connect to asterisk.
 Is there anybody who knows why (and maybe how to solve it)?

I'll say this: IMHO safe_asterisk reduces stability. This is because it
can cause in illussion of asterisk being up whereas asterisk is
actually down.

Is anybody actually experincing frequent crashes? daily? weekly?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Re: Cisco 79xx and vlan

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 I know that this has been asked before, but I couldn't find an answer ..
 
 In the office, my 79XX phones (connected to dell / hp switches) are all 
 on their own separate network (i.e. we have data going through separate 
 switches). When they boot, they take ages on the configuring VLAN screen.
 
 However, I also have a 7960 at home, connected to work through a vpn. 
 This one boots very quickly indeed. It's not the phone settings, as I 
 took this phone into the office and it then had the same symptoms.
 
 Has anyone got any idea on how to speed this process up ?
 
 On a side note, does anyone know how to send a reload config command 
 to the 7940 without having to reboot it ?

Hi Julian!

I have several Cisco phones and I'm interested to get answers to your 
questions. If you find solution, please send mail to the list.

P.S.
Are Cisco phones able to do paging/intercom?

--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Getting 'i' functionality on internal extensions

On Thu, Sep 14, 2006 at 10:23:09AM -0500, Eric ManxPower Wieling wrote:
 exten = _X.,1,Playback(pbx-invalid)
 exten = _X.,2,Goto(s,1)

The problem with this is that all extensions now take 3 seconds longer to
answer. For example, with this extensions.conf:

[internal]
exten = 611,1,Answer()
exten = 611,2,Playback(hello-world)
exten = 611,3,Hangup()
exten = _X.,1,Answer()
exten = _X.,2,Playback(pbx-invalid)
exten = _X.,3,Hangup()

After dialling 611 there is a three-second pause as Asterisk tries to
match more digits, before it decides to connect.

Without the last three lines, it connects immediately, and also detects as
soon as the pattern can never possibly match - e.g. if you dial 62 or 8.
However, in this case what I get is a busy tone on my phone (connected to
FXS port on TDM400P), as if the number I dialled was correct but the far end
was busy. That's not very helpful to users.

So if I could select a recorded message to play - or even just change the
tone to Number Unobtainable - that would be much better.

Hmm... digs around... pattern _X! seems to be what I want. It even seems to
work if I also have wildcard _9. to dial outgoing lines.

OK, problem solved - thanks for pointing me in the right direction!

Regards,

Brian.
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[asterisk-users] Anyone using Voicemail with IMAP Support?

2006-09-15 Thread Arnd Vehling


Hi,

ive tried to setup a svn trunk version of asterisk to test
voicemail with imap support and i am so far without success.

Is there _anyone_ running voicemail with IMAP Support who can
answer some basic questions?

regards,

  Arnd

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[asterisk-users] Re: Can you explain why multiple registration is an important (missing) feature ?

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 In some cases : Yes.
 But we have the following situation : We re using cisco 7960 phones in
 each office (about 150 of them)

Do Cisco phones support paging/intercom? If yes, please send me link to some 
useful pages.

 Now we want to give the user's the ability to take their number with
 them. So when you change places you can call a defined number which
 will write you a config file for your new phone.

To much work. Is it working right?

 Now, if I have extension 1234 and go to a different office, or to a
 meeting room, etc and log into that phone using my extension, if i did
 not log out my normal phone we have a problem because we have to SIP/1234.
 I haven't found a good solution for that yet, but if I could register
 two SIP/1234 phones the problem would be solved.

I would like that Asterisk supports multiple registers, but till then you could 
use dynamic agents. Agent can log in from every phone. And you send incoming 
phone call to agent instead to extension.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] two safe_asterisk processes on the same PBX???


Hi Tzafrir,
safe_asterisk use was encouraged on wiki pagesit has never given me 
crash problems or something similarthe problem I have is to have two 
safe_asterisk processes which causes a lot of  messages inside logs.


How did you replace safe_asterisk?

Giorgio.


Tzafrir Cohen wrote:

On Fri, Sep 15, 2006 at 09:52:02AM +0200, Giorgio Incantalupo wrote:
  

Hi,
I noticed sometimes I get the messages remote unix connection every 
1or 2 seconds. I found that there is a second safe-asterisk process 
which is probably trying to start/connect to asterisk.

Is there anybody who knows why (and maybe how to solve it)?



I'll say this: IMHO safe_asterisk reduces stability. This is because it
can cause in illussion of asterisk being up whereas asterisk is
actually down.

Is anybody actually experincing frequent crashes? daily? weekly?

  


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Re: [asterisk-users] voicemail access thru apache on another server

2006-09-15 Thread Arnd Vehling

Hi Benjamin,

Am trying to build a system, wherein users can access their profiles,
and hence voicemails thru a browser.
I am using Apache and am running it on another box and asterisk on
another. Am keeping them seperate to not have http traffic on the same
box as asterisk.

Now, my qs:
Is there a way to tell Asterisk to store the msg.txt information in
an sql database, so that it's easier to access the voice mail info??

I would wait until the IMAP support in asterisk (currently only in the
svn version) is stable instead using sql-db based storage. At least if
were talking about an medium installation or bigger.

Also, any way to run a script or something, to move a message from INBOX
to Old, when a user listens to the message thru the web browser??

Now, how on earth do i read the recordings and play them out thru a browser!!

I did write a proof of concept script in php for accessing and manipulating
the voicemail folder. It has no locking atm therefore there are some race
conditions and its not usuable in a large scale production environment.

Youre welcome to add session/locking suport though :}
The script enables you to view, read, move, delete and forward voicemails
using URLs. ascii based and asterisk realtime authentication is supported.
You can use it at least to extract the code how to send an audiofile (if
youre using php that is). Should be no problem to convert the scripts
to perl or any other script language.

Script:
http://sip-syndication.com/index.php?option=com_remositoryItemid=26func=selectid=2
Documentation:
http://sip-syndication.com/index.php?option=com_contenttask=categorysectionid=5id=21Itemid=47

cheers,

Arnd
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Re: [asterisk-users] How to download asterisk 1.3 development version

On Fri, Sep 15, 2006 at 04:10:30PM +1000, Boneym wrote:
 I would like to test asterisk 1.4 development version , can anyone
send me a link to it . Thanks in advance.

Try this:

(1) Open your web browser
(2) Enter www.asterisk.org
(3) Click on the link marked downloads, at the top of the page
(4) Scroll down to where it says SVN repository
(5) Follow the instructions to download the trunk source
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[asterisk-users] trying to understand siprealtime nat/MWI issues

2006-09-15 Thread Arne Van Theemsche

Hi

I am trying to understand why, if you don't use realtime users caching, the NAT  MWI doesn't work with realtime sipfriends.

Is it because the code responsible for that doesnt work with realtime yet, or is there another thing I am missing? 

thanks
Arne

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Re: [asterisk-users] How to download asterisk 1.3 development version

2006-09-15 Thread Paul Hales


BoneyM  - your best bet is to download Asterisk from the SVN repository
(also known as trunk) - details are on the downloads page at
www.asterisk.org

later,

PaulH
AsteriskIT


On Fri, 2006-09-15 at 16:10 +1000, Boneym wrote:
 Hi,
 
  I would like to test asterisk 1.4 development version , can anyone
 send me a link to it . Thanks in advance.
 
  
 
 Cheers,
 
 boneyM
 
 
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[asterisk-users] 491 request pending [2]

2006-09-15 Thread harrygaillac-sip

Hello,


Why 491 pending though asterisk send INVITE to ser
proxy ?

How may I setup this configuration ?


here Is my Problem:

I want asterisk to sent none local URI to SER  

My config asterisk svn-trunk:

UA===SER=ASTERISK===SER===sip URI
---INVITEINVITE-INVITE---
491-491 req pending

I set a peer with outboundproxy so in extensions.conf
I forward to ser non local URI .

Any Idea

harry








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Re: [asterisk-users] two safe_asterisk processes on the same PBX???

2006-09-15 Thread Julian Lyndon-Smith


Giorgio Incantalupo wrote:

Hi Tzafrir,
safe_asterisk use was encouraged on wiki pagesit has never given me 
crash problems or something similarthe problem I have is to have two 
safe_asterisk processes which causes a lot of  messages inside logs.


This is because you already have an asterisk process running. When you 
run safe_asterisk, it fails, and attempts to restart.


Either

A) asterisk -r; stop now
B) Kill the safe_asterisk process that is attempting to restart



How did you replace safe_asterisk?

Giorgio.


Tzafrir Cohen wrote:

On Fri, Sep 15, 2006 at 09:52:02AM +0200, Giorgio Incantalupo wrote:
 

Hi,
I noticed sometimes I get the messages remote unix connection every 
1or 2 seconds. I found that there is a second safe-asterisk process 
which is probably trying to start/connect to asterisk.

Is there anybody who knows why (and maybe how to solve it)?



I'll say this: IMHO safe_asterisk reduces stability. This is because it
can cause in illussion of asterisk being up whereas asterisk is
actually down.

Is anybody actually experincing frequent crashes? daily? weekly?

  


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Julian
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[asterisk-users] Setting up imap based voicemail / invalid remote specification

2006-09-15 Thread Arnd Vehling


Hi,

ive just installed a svn trunk (r42858) and i am having problems
getting app_voicemail to even try to connect to a imap server.

Ive added the following to voicemail.conf
--
; new IMAP Stuff
imapserver=mydom.com
imapport=143
expungeonhangup=no
[..]
[default]
; Office Accounts
7709810 = 1234, Me Myself, [EMAIL 
PROTECTED],,attach=yes|imapuser=me|authuser=me
|autpassword=mypass
--

When trying to leave a voicemail ill get:
--
[Sep 13 16:20:45] ERROR[30445]: app_voicemail.c:8193 mm_log: IMAP Error: Can't
open mailbox {mydom.com:143/imap//user=me}INBOX: invalid remote specification
[Sep 13 16:20:45] ERROR[30445]: app_voicemail.c:2455 count_messages_imap:
Houston we have a problem - IMAP mailstream is NULL
[Sep 13 16:20:45] NOTICE[30445]: app_voicemail.c:2876 leave_voicemail: Can not
leave voicemail, unable to count messages
--

Any hints?

BTW: If i call Voicemail with Voicemail([EMAIL PROTECTED]|sb) in
extension.conf the ${CONTEXT} gets replaced with the actual context but
asterisk still tries to find the mailbox in the default context? Am i
missing something here (Docu for changed extension.conf syntax? = where?)
or is this a bug?

regards,

  Arnd
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Re: [asterisk-users] Re: Can you explain why multiple registration is an important (missing) feature ?

2006-09-15 Thread Ricardo Carvalho

You can use Asterisk along with Ser. Asterisk for advanced features like 
Voicemail and gateway, and Ser for routing SIP messages, Registrar, acc, 
etc. Take a look at:

http://www.voip-info.org/wiki-Asterisk+at+large
It works!!

Regards,
Ricardo.




Tomislav Parčina wrote:

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
  

In some cases : Yes.
But we have the following situation : We re using cisco 7960 phones in
each office (about 150 of them)



Do Cisco phones support paging/intercom? If yes, please send me link to some 
useful pages.

  

Now we want to give the user's the ability to take their number with
them. So when you change places you can call a defined number which
will write you a config file for your new phone.



To much work. Is it working right?

  

Now, if I have extension 1234 and go to a different office, or to a
meeting room, etc and log into that phone using my extension, if i did
not log out my normal phone we have a problem because we have to SIP/1234.
I haven't found a good solution for that yet, but if I could register
two SIP/1234 phones the problem would be solved.



I would like that Asterisk supports multiple registers, but till then you could 
use dynamic agents. Agent can log in from every phone. And you send incoming 
phone call to agent instead to extension.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] voicemail access thru apache on another server

2006-09-15 Thread Tzafrir Cohen

On Thu, Sep 14, 2006 at 06:14:52PM +0530, Benjamin Jacob wrote:
 Hello ppl,
 Am trying to build a system, wherein users can access their profiles, 
 and hence voicemails thru a browser.
 I am using Apache and am running it on another box and asterisk on 
 another. Am keeping them seperate to not have http traffic on the same 
 box as asterisk.

Just to give you another direction: mod_proxy of apache and a locla
httpd on the box that runs asterisk...

 
 Now, my qs:
 Is there a way to tell Asterisk to store the msg.txt information in 
 an sql database, so that it's easier to access the voice mail info??

Or use the mysql, postgresql or ODBC storage.

Or use the imap storage and some web-based imap client.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] 9 becomes 99 ? And other strangeness

On Thu, Sep 14, 2006 at 08:33:43PM -0500, Eric ManxPower Wieling wrote:
 Turn off relaxdtmf in zapata.conf if that does not help play with the 
 rxgain, if that does not help, play with the txgain.  If the volume is 
 too loud or too soft on zap channels, Asterisk can sometimes miss or see 
 double DTMF.

That's the pointer I needed.

With lots of experimentation, it was clear that double DTMF only occurred on
the first digit - and this is while the dial tone is being played, of
course.

Setting txgain=-6.0 seemes to have solved this.

Many thanks,

Brian.
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Re: [asterisk-users] two safe_asterisk processes on the same PBX???

2006-09-15 Thread Tzafrir Cohen

On Fri, Sep 15, 2006 at 11:02:37AM +0200, Giorgio Incantalupo wrote:
 Hi Tzafrir,
 safe_asterisk use was encouraged on wiki pagesit has never given me 
 crash problems or something similarthe problem I have is to have two 
 safe_asterisk processes which causes a lot of  messages inside logs.
 
 How did you replace safe_asterisk?

Just run asterisk as a daemon. There is simply no need in an extra
wrapper script. I have an init.d script anyway.

I did ad a debug option to the debian init.d script to start asterisk
undaemonized (but with all the other options. Most notbly -U and -p) t
ohelp me trace issues where asterisk does not start. This is mainly
because people (me included) tend to forget to add -U and then a simple
run of 'asterisk -vv' ends up with root-owned files that
prevent a proper asterisk from starting the next time.

As I wrote: the scrpt safe_aterisk has originated, from my undertnading,
as a workaround to a problem of Asterisk crashing. IMHO you should use
it as a temporary workaround if you already have experinced crashes and
want to keep PBX up while tracking down the issue. But first and
formost, you should fix the issue.

safe_aterisk comes at the price of complicating the setup. As in your
case. And it is not a silver-bullet, either. put one bogus module in
/usr/lib/modules/asterisk . This will cause asterisk to segfault at
startup. With safe_asterisk you will get a crash-loop.

With an init.d script the script will return success (as it has not
failed before daemonizing) but at least you won't have a service
running.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Modem calls

2006-09-15 Thread Jose Limeres

Hi,
I need to pass modem calls through a TDM400 card. Conecting the modem
to the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4)
directly.
Even though Echo cancellation is disabled in both lines the call is
never successful. Modems speak for some time and then the line is hang
up with the message No answer from remote side. I send thereafter
some traces in case someone is interested.

I know modem calls are not easy as something is messed up in the
signaling going through asterisk but has someone been successful in an
easy configuration such as this one?

Thanks,
Jose
Sep 15 09:45:22 DEBUG[4625] chan_zap.c: Dialing '908274101'
Sep 15 09:45:22 DEBUG[4625] chan_zap.c: Deferring dialing...
Sep 15 09:45:22 DEBUG[4625] chan_zap.c: Requested indication 3 on channel 
Zap/1-1
Sep 15 09:45:23 DEBUG[4625] chan_zap.c: Exception on 15, channel 4
Sep 15 09:45:23 DEBUG[4625] chan_zap.c: Got event Hook Transition Complete(12) 
on channel 4 (index 0)
Sep 15 09:45:25 DEBUG[4625] chan_zap.c: Exception on 15, channel 4
Sep 15 09:45:25 DEBUG[4625] chan_zap.c: Got event Dial Complete(9) on channel 4 
(index 0)
Sep 15 09:45:25 DEBUG[4625] chan_zap.c: No echo cancellation requested
Sep 15 09:45:25 DEBUG[4625] chan_zap.c: No echo training requested
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Exception on 15, channel 4
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Got event Dial Complete(9) on channel 4 
(index 0)
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: No echo cancellation requested
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Requested indication -1 on channel 
Zap/1-1
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Took Zap/1-1 off hook
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: master: 1, slave: 4, nothingok: 0
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Stopping tones on 1/0 talking to 4/0
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Stopping tones on 4/0 talking to 1/0
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Making 4 slave to master 1 at 0
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Added 15 to conference 9/1
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Added 13 to conference 9/4
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Updated conferencing on 1, with 0 
conference users
Sep 15 09:45:27 DEBUG[4625] chan_zap.c: Updated conferencing on 4, with 0 
conference users
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Exception on 13, channel 1
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Got event On hook(1) on channel 1 
(index 0)
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: No echo cancellation requested
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: No echo cancellation requested
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Unlinking slave 4 from 1
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Removed 15 from conference 9/1
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Removed 13 from conference 9/4
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Updated conferencing on 1, with 0 
conference users
Sep 15 09:45:51 DEBUG[4625] channel.c: Returning from native bridge, channels: 
Zap/1-1, Zap/4-1
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Hangup: channel: 4 index = 0, normal = 
15, callwait = -1, thirdcall = -1
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Set option TDD MODE, value: OFF(0) on 
Zap/4-1
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Updated conferencing on 4, with 0 
conference users
Sep 15 09:45:51 DEBUG[4625] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Sep 15 09:45:51 DEBUG[4625] cdr_addon_mysql.c: cdr_mysql: inserting a CDR 
record.
Sep 15 09:45:51 DEBUG[4625] cdr_addon_mysql.c: cdr_mysql: SQL command as 
follows: INSERT INTO cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
 VALUES ('2006-09-15 09:45:22','\FXS\ 
505','505','0908274101','from-internal', 
'Zap/1-1','Zap/4-1','Dial','ZAP/4/908274101|120|r',29,24,'ANSWERED',3,'','asterisk-3279-1158306315.0')
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Hangup: channel: 1 index = 0, normal = 
13, callwait = -1, thirdcall = -1
Sep 15 09:45:51 DEBUG[4625] chan_zap.c: Set option TDD MODE, value: OFF(0) on 
Zap/1-1


asterisk1*CLI zap show channel 1
Channel: 1CLI
File Descriptor: 13
Span: 1
Extension:
Dialing: no
Context: from-internal
Caller ID: 505
Calling TON: 0
Caller ID name: FXS
Destroy: 0
InAlarm: 0
Signalling Type: FXO Kewlstart
Radio: 0
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 0 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Onhook
asterisk1*CLI

asterisk1*CLI
asterisk1*CLI
asterisk1*CLI zap show channel 4
Channel: 4CLI
File Descriptor: 15
Span: 1
Extension:
Dialing: no
Context: from-zaptel
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no

[asterisk-users] CDR question with SIP/IAX trunks

2006-09-15 Thread Koopmann, Jan-Peter

Hi,

scenario:

Call comes in via ISDN BRI on Asterisk A. Callerid (set by zapata) is let's say 
0151123456789. In the incoming context I prepend a 0 to that callerid. My snom 
correctly displays 00151123456789. The call is also forwarted to Asterisk B. 
On the incoming context of Asterisk B I prepend yet another prefix 98. The 
callerid now is 9800151123456789 which is correctly displayed on the SNOM on 
Asterisk B. So far so good.

The CDR on Asterisk B logs the callerid 001511234567890 though and ignores the 
prepended 98. Any ideas why?

Kind regards,
  JP
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Re: [asterisk-users] two safe_asterisk processes on the same PBX???


Hi Julian,
I know I have two process.the problem is I launch only one but 
sometimes (I do not when or why) another process is launched.
Tzafrir Cohen told me to avoid safe_asterisk..I'll think about it and 
then create my own launch script.





Giorgio Incantalupo

Julian Lyndon-Smith wrote:

Giorgio Incantalupo wrote:

Hi Tzafrir,
safe_asterisk use was encouraged on wiki pagesit has never given 
me crash problems or something similarthe problem I have is to 
have two safe_asterisk processes which causes a lot of  messages 
inside logs.


This is because you already have an asterisk process running. When you 
run safe_asterisk, it fails, and attempts to restart.


Either

A) asterisk -r; stop now
B) Kill the safe_asterisk process that is attempting to restart



How did you replace safe_asterisk?

Giorgio.


Tzafrir Cohen wrote:

On Fri, Sep 15, 2006 at 09:52:02AM +0200, Giorgio Incantalupo wrote:
 

Hi,
I noticed sometimes I get the messages remote unix connection 
every 1or 2 seconds. I found that there is a second safe-asterisk 
process which is probably trying to start/connect to asterisk.

Is there anybody who knows why (and maybe how to solve it)?



I'll say this: IMHO safe_asterisk reduces stability. This is because it
can cause in illussion of asterisk being up whereas asterisk is
actually down.

Is anybody actually experincing frequent crashes? daily? weekly?

  


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Julian
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RE: [asterisk-users] CDR question with SIP/IAX trunks

2006-09-15 Thread Koopmann, Jan-Peter

Sorry for replying to my own post: I just switch the connection from Asterisk A 
to Asterisk B from SIP to IAX without changing anything else (dialplans on both 
system are the same). Now the correct callerID is logged. The behaviour changed 
from 1.2.9 to 1.2.10 I suppose since this worked without problems in August. 
After my switch to the new version, this started going wrong...

Any ideas?
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[asterisk-users] Compile error in Asterisk 1.2.12.1

2006-09-15 Thread Henrik Woffinden

Hi.

I'm using zaptel-1.2.9.1/libpri-1.2.3/asterisk-1.2.12.1 all patched with
bristuff-0.3.0-PRE1s.

What could be the problem when I get this compiler error:
-- cut ---
gcc  -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3  -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE  -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-fomit-frame-pointer   -DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC   -c -o
res_agi.o res_agi.c
res_agi.c: In function 'agi_exec_full':
res_agi.c:2120: error: too few arguments to function 'launch_script'
res_agi.c:2124: error: 'AGI' has no member named 'audio'
res_agi.c:2094: warning: unused variable 'efd2'
make[1]: *** [res_agi.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-1.2.12.1/res'
make: *** [subdirs] Error 1
-- cut ---

-- 
Med venlig hilsen / Best regards,

Henrik Woffinden
Technical Director
Nitram Lexa ApS
Maglebjergvej 5A
DK-2800 Kongens Lyngby
Denmark

Phone: +45 70 25 24 23 Fax: +45 70 25 29 23
Mobile: +45 40 85 25 17

E-mail: [EMAIL PROTECTED] Web: www.nitramlexa.com

---
Windows is a 32-bit extension to a 16-bit graphical shell for an 8-bit
operating system originally coded for a 4-bit microprocessor by a 2-bit
company that can't stand 1 bit of competition.

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Re: [asterisk-users] mISDN versus ZapHFC with BRIstuff

2006-09-15 Thread Kai Ober


is it a single s0 card?
how do you ring the 3 phones?

no problems with the installation of mISDN so far.

it is as easy as on Bristuff


regards
KAI


Henrik Woffinden schrieb:

Hi

Sorry... I haven't been specific enough...

I have several ISDN phones on my inside NT mode ISDN card, and I wan't
3 of the MSN (local) numbers to ring at the same time. I can't get more
than 2 phones to ring at the same time, unless I ring them all by
dialing the group, but that's not what I want.
  


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Re: [asterisk-users] non-technical, dealing with users giving feedback

2006-09-15 Thread stoffell


Search Daily Asterisk News for echo:


Yes, that's for the issue with echo, but I was more or less meaning
the social side, the communication with the users.. echo was an
example.. :) (bad choice maybe? :))

cheers
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Re: [asterisk-users] WAIT FOR DIGIT not working

2006-09-15 Thread Kai Ober


prints print really to stdout?,
flushed the output?



$target = ;

print WAIT FOR DIGIT 5000\n;
$target .= STDIN;
print WAIT FOR DIGIT 5000\n;
$target .= STDIN; 
print WAIT FOR DIGIT 5000\n;

$target .= STDIN;

  


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[asterisk-users] Cisco Distinctive ring using alert-info

2006-09-15 Thread Julian Lyndon-Smith

I've got a cisco 7960, with (amongst many others) the following in the 
RINGLIST.DAT file


Foghorn foghorn.raw

I can manually select this for the ringtone. However, I was wanting to 
use a normal ringtone, with foghorn being used if the call was coming in 
from the girlfriend/wife/mother-in-law etc ;)


I was trying to use the following:

exten = 5711,1,SIPAddHeader(Alert-Info: Foghorn)
exten = 5711,n,Dial(SIP/5711)
exten = 5711,n,Hangup()

However, not matter what I try, I get the standard ringtone. If I use

exten = 5711,1,SIPAddHeader(Alert-Info: Bellcore-dr3)

then I get a different ringtone.

Can you tell the 7960 to play a certain ringtone, or are you limited to 
a certain set ?


Julian
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Re: [asterisk-users] callback without agi

2006-09-15 Thread Kai Ober





exten = 333,n,Authenticate(1234)
.
.
exten = 333,n+101,NoOp(Is this ok??)


Or i have to explicitly enumerate the priority? ... i'm searching for 
doc about this.
as far as i know Auth( ) does not jump to n+101 if you dont use 
Auth..(123,j)


enumrations are easier if you use somthing like  Goto(s,4) , with n you 
dont know where you wanna go.


regards
KAi

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Re: [asterisk-users] non-technical, dealing with users giving feedback

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

stoffell wrote:
 Search Daily Asterisk News for echo:
 
 Yes, that's for the issue with echo, but I was more or less meaning
 the social side, the communication with the users.. echo was an
 example.. :) (bad choice maybe? :))

:)

I kinda knew that you meant that, but was saying that pretty much all
issues can be resolved, you just need to search for the solutions, give
them a tracking number, and let them know you're working on a fix.

Update the customer on the progress, and if possible let them know how
long you expect it to take to resolve.

Ideally you will have a support contract with them, including a certain
number of hours per month, and if it runs over that then charge per hour.

If you don't find the solution to your problem via google, feel free to
ask here, and hopefully someone will help you.

If a feature is required that doesn't exist, post a bounty (hopefully
your support contract will cover it, but if not let the customer know,
and pass on the charge).

After the feature has been completed, submit it to the bugtracker
(bugs.digium.com) so that the rest of the community can benefit from the
change.

If everyone does that, we all benefit.

:)

- --
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFFCpXjS6d5vy0jeVcRAmi/AJ9CT9wWeRYbbK8tA7pXT6unUbzs+gCaAlAh
L6SOoRmyL2u4krstMQwLDBU=
=3Wqy
-END PGP SIGNATURE-
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Re: [asterisk-users] 9 becomes 99 ? And other strangeness

2006-09-15 Thread Andrew Kohlsmith

On Friday 15 September 2006 04:20, Brian Candler wrote:
 it worse: I got a failure rate of about 50%, and in one case 66611 instead
 of 611.

It's clear your system is possessed.  Please contact your local clergyman for 
help with these issues.

-A.

(seriously though, I've had this particular problem on one box.  Properly 
tuning the tx/rx gains for zapata helped eliminate this problem.)

Interestingly enough, I tuned my rxgain for 14844, but my *transmit* audio 
levels had to be around half that before the echo cancellers were AT ALL 
happy.

-A.
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Re: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls

2006-09-15 Thread Frederik Fix

Just tried it. When I run sip show channels it doesnt show any open  
channels.


Thanks,
Frederik

On 14 Sep 2006, at 03:27, Bill Gibbs wrote:


Make those calls then check the CLI sip show channels and see if the
channels are stay up

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Frederik
Fix
Sent: Wednesday, September 13, 2006 8:35 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working
togetherafter 3 calls

Hi,
I have a strange problem that I have no idea how to debug:

I have a Zyxel Prestige 2000W Wifi telephone that is connected to my
Asterisk server which has a Junghanns.net QuadBRI card. I can make
exactly 3 calls to the outside over the QuadBRI. Any calls after
that fail with the log saying that all lines are busy.

Turning the phone off and on solves the problem and I can make 3
calls again before it repeats. This problem does not occur when I
make calls from my Cisco 7960G phones using SCCP or using eyebeam and
SIP. Also making calls from the Zyxel through a cheap Cologne chipset
ISDN card using zaphfc does not show this problem.

I am using the following versions:
Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r
Zyxel Prestige 2000W (version 1)
Zyxel-Firmware: Wj.00.11


Any help is very much appreciated,

Frederik

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RE: [asterisk-users] CDR question with SIP/IAX trunks

2006-09-15 Thread Koopmann, Jan-Peter

Ok... I got it. Someone changed the CDRs to reflect CALLERID(ANI) instead of 
CALLERID(number) in 1.2.10. According to the release notes this was taken back 
in 1.2.12. I do not know why this was not done for IAX as well so it would have 
been consistent at least but well... 


I am either going to set ANI now or upgrade to 1.2.12... :-)
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Re: [asterisk-users] RE: Asterisk 1.4 Docs

2006-09-15 Thread Jason A. Kates

One of the providers that I use already offers this feature via a macro
in the dail plan
http://connect.voicepulse.com/FlexRate.aspx

-Jason

On Fri, 2006-09-15 at 10:21 +0200, Tomislav Parčina wrote:
 In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
  No mention of Shared Line Appearance in the v1.4 new release.  Anyone know
  if they still plan to include it or not?  Digium has been kind of quiet on
  their work on that feature.
  
  With their new Asterisk appliance running v1.4 I certainly hope they have
  SLA as all other traditional/proprietary PBX's in that market segment do.
 
 Yes, and I'm interested in AOC messages. If I'm only able to manipulate with 
 them, store them somewhere.
 
 I believe every Asterisk user will benefit with this, it just that people are 
 not familiar what AOC does. AOC messages (Advice of charge) are messages that 
 your provider sends you at the end of call. They tell you how much units jour 
 provider will charge you for that call. And if you would like to know how 
 much money is that, you simply multiply it with price of every unit.
 
 It will solve charging problems with Asterisk! We wouldn't have to keep up to 
 date our databases with prices. Provider will directly tell us how much he 
 will charge every our call.
 
 How to help/motivate developers to work on AOC in Asterisk?
 
 
 --
 Tomislav Parčina
 Lama Computers Split
 Stinice 12, 21000 Split
 Tel.: +385(21)495148
 Mob.: +385(91)1212148
 SIP: [EMAIL PROTECTED]
 e-mail: tparcina#lama.hr
 http://www.lama.hr
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-- 

Jason A. Kates ([EMAIL PROTECTED]) 
Fax:208-975-1514
Phone:  212-400-1670 x2


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Re: [asterisk-users] Modem calls

2006-09-15 Thread Time Bandit


 I need to pass modem calls through a TDM400 card. Conecting the modem to
the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4)
directly.

According to Digium, Fax calls (and modem calls) are not supported on
the TDM400 or TDM2400. They are designed for voice only. If you get it
to work, you're lucky.

Sangoma test all their cards with faxes, so maybe you should try their card.

For your problem, run zttest and adjust everything to try to obtain
100%, this may make it work.

hth
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[asterisk-users] Section '12345678' lacks type

2006-09-15 Thread Lennart Utgård




I'm testing the use of static 
SQL-config. Everything seems to work OK, exept these warnings: 
Sep 15 15:06:20 WARNING[25374]: chan_sip.c:12829 reload_config: Section 
'157217030' lacks typeSep 15 15:06:20 WARNING[25374]: chan_sip.c:12829 
reload_config: Section 'TISP-157217030' lacks typeSep 15 15:06:20 
WARNING[25374]: chan_sip.c:12829 reload_config: Section '157217030' lacks 
typeSep 15 15:06:20 WARNING[25374]: chan_sip.c:12829 reload_config: Section 
'TISP-157217030' lacks typeSep 15 15:06:20 WARNING[25374]: chan_sip.c:12829 
reload_config: Section '157217030' lacks typeSep 15 15:06:20 WARNING[25374]: 
chan_sip.c:12829 reload_config: Section 'TISP-157217030' lacks typeThis 
occurs even if the 'type' variable is set for the sections. 

SELECT * FROM 
`ast_config` WHERE `var_name` LIKE 'type' filename category var_name 
var_val sip.conf TISP-157217030 type peer sip.conf 157217030 type friend 

Everything seems to 
work.

Is it a bug? Any idea, anyone?
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Re: [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?

2006-09-15 Thread Eric \ManxPower\ Wieling


Christian Mohrbacher wrote:

In some cases : Yes.
But we have the following situation : We re using cisco 7960 phones in
each office (about 150 of them), but not every person has it's own
phone. Normally there are two employees in one office and they share one
phone, BUT have their own extension. Fortunately the Ciscos are able to
register each line seperately. The lines are registered via their
extension, which means if you configure your phone to have extension
1234 the line will be registeres as SIP/1234.


And there is your problem.  Using the extension as the SIP User ID does 
not scale, is confusing, and limits your thinking about devices and 
extensions.  There are several reasons this is a bad idea.  Multiple 
extension numbers ringing on the same device / line appearance is the 
most common.


We use the MAC address of the device as the SIP User ID.  We append a 
-a, -b, -c, etc to the MAC address for each line appearance.  This does 
not work well for Softphone, but since All Softphones Suck(TM), we don't 
really care about this limitation.


Users seldom need to know their SIP User ID.
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[asterisk-users] where download app_txfax?

2006-09-15 Thread Jerry Geis


I have gone to http://soft-switch.org/downloads/spandsp/
looking to app_rxfax and I dont see it? Where is it?

Parent Directory http://soft-switch.org/downloads/ -   
[TXT] app_dtmftotext.c http://soft-switch.org/downloads/spandsp/app_dtmftotext.c17-Mar-2004 03:18   28K  
[DIR] spandsp-0.0.2pre20/ http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre20/ 05-Apr-2006 12:43-   
[DIR] spandsp-0.0.2pre21d/ http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21d/05-Apr-2006 12:43-   
[DIR] spandsp-0.0.2pre23/ http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre23/ 05-Apr-2006 12:43-   
[DIR] spandsp-0.0.2pre25/ http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre25/ 21-May-2006 12:10-   
[DIR] spandsp-0.0.2pre26/ http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/ 28-May-2006 07:12-   
[   ] spandsp-0.0.3pre6.tgz http://soft-switch.org/downloads/spandsp/spandsp-0.0.3pre6.tgz   31-Mar-2006 10:46  1.3M  
[   ] spandsp-0.0.3pre20.tgz http://soft-switch.org/downloads/spandsp/spandsp-0.0.3pre20.tgz  10-Jun-2006 09:11  1.4M  
[   ] spandsp-0.0.3pre21.tgz http://soft-switch.org/downloads/spandsp/spandsp-0.0.3pre21.tgz  12-Jun-2006 22:47  1.4M  
[   ] spandsp-0.0.3pre22.tgz http://soft-switch.org/downloads/spandsp/spandsp-0.0.3pre22.tgz  26-Jun-2006 23:23  1.4M  


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Re: [asterisk-users] Astmanproxy authentication problems

2006-09-15 Thread Leonardo Gomes Figueira

Fabian,

[EMAIL PROTECTED] escreveu:
 I've try to use Astmanproxy with Asterisk TAPI line.
 But login fails,  astmanproxys error message:
 
 Sep 13 20:06:26: [EMAIL PROTECTED] got: Response: Error
 Sep 13 20:06:26: [EMAIL PROTECTED] got: Message: No variable specified
 Sep 13 20:06:26: [EMAIL PROTECTED]: attempting read...

You should capture the tcp stream with tcpdump/ethereal or other tool
and post the text session on the astmanproxy list:

http://groups.yahoo.com/group/asterisk-astmanproxy/

So we can help you there.

  Leonardo

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[asterisk-users] Issues with AGI+Dial command

2006-09-15 Thread Brian Rogan

Hello,

I am trying to write an AGI application that will transfer the caller to
a phone number on certain conditions.  From what I understand (from the
astcc application and voip-info wiki), I should just be able to EXEC the
dial command.  I'm having problems with this though.  I send asterisk
the following:

EXEC Dial Zap/g1/8475881188|30

I get back:

200 result=-1

On the asterisk console I see:

-- AGI Script Executing Application: (Dial) Options:
(Zap/g1/8475881188|30)
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/8475881188
-- Hungup 'Zap/1-1'


The HANGUPCAUSE variable is set to 0.  When I put this dial in my
dialplan as Dial(Zap/g1/8475881188|30), the call goes through fine, so I
don't think that its the T card or any configuration.

Has anyone ever seen this before?

Thanks,

--Brian
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Re: [asterisk-users] Re: Streaming MoH Problem, starts and then stops immediately

CentOS 3.8, Asterisk 1.2.9.1, AMD Sempron(tm) Processor 3000+, mpg123-0.59r
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Re: [asterisk-users] Re: chan_zap.so stopped working after upgrading CentOS

Why zaptel 1.2.5 and not the newer version?
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Re: [asterisk-users] How to install HUDLite Server

HUDLite has a very impressive interface. Works fine for me on Trixbox, but couldn't make it work on standalone Asterisk.

What kinds of bugs are there in it? Has anybody used its full version, how is it. I am thinking of upgrading one of my clients to fonality, just because of HUD's interface and easy to operate options, as they show in its demo.

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RE: [asterisk-users] Sphinx2

2006-09-15 Thread Douglas Garstang

What sphinx documentation? All I could find was docs on the code, not on how to 
USE the software.

 -Original Message-
 From: Matt Riddell (IT) [mailto:[EMAIL PROTECTED]
 Sent: Friday, September 15, 2006 1:18 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Sphinx2
 
 
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 Douglas Garstang wrote:
  The docs at that URL say that the dictionary has 'yes' in 
 it... although I don't understand how I can get replies like 
 'YOU HALF' if it doesn't exist in the dictionary.
 
 Did you read the Sphinx documentation?
 
 Rather heavy but describe everything!
 
 - --
 Cheers,
 
 Matt Riddell
 ___
 
 http://www.sineapps.com/news.php (Daily Asterisk News - html)
 http://wap.sineapps.com (Daily Asterisk News for your cellphone)
 http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.2 (MingW32)
 Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
 
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Re: [asterisk-users] two safe_asterisk processes on the same PBX???

2006-09-15 Thread Moises Silva


If you want to have a safe asterisk I would recommend using svscan
from daemontools package, more wonderfull software of D.J. Bernstein.

http://cr.yp.to/daemontools/svscan.html

Regards

On 9/15/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:

Hi Julian,
I know I have two process.the problem is I launch only one but
sometimes (I do not when or why) another process is launched.
Tzafrir Cohen told me to avoid safe_asterisk..I'll think about it and
then create my own launch script.




Giorgio Incantalupo

Julian Lyndon-Smith wrote:
 Giorgio Incantalupo wrote:
 Hi Tzafrir,
 safe_asterisk use was encouraged on wiki pagesit has never given
 me crash problems or something similarthe problem I have is to
 have two safe_asterisk processes which causes a lot of  messages
 inside logs.

 This is because you already have an asterisk process running. When you
 run safe_asterisk, it fails, and attempts to restart.

 Either

 A) asterisk -r; stop now
 B) Kill the safe_asterisk process that is attempting to restart


 How did you replace safe_asterisk?

 Giorgio.


 Tzafrir Cohen wrote:
 On Fri, Sep 15, 2006 at 09:52:02AM +0200, Giorgio Incantalupo wrote:

 Hi,
 I noticed sometimes I get the messages remote unix connection
 every 1or 2 seconds. I found that there is a second safe-asterisk
 process which is probably trying to start/connect to asterisk.
 Is there anybody who knows why (and maybe how to solve it)?


 I'll say this: IMHO safe_asterisk reduces stability. This is because it
 can cause in illussion of asterisk being up whereas asterisk is
 actually down.

 Is anybody actually experincing frequent crashes? daily? weekly?



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 Julian
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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [asterisk-users] How to install HUDLite Server