date:20040901

Norman Tomlnis [EMAIL PROTECTED] wrote:
> I am trying to do a make clean; make install in the asterisk source
> directory and I am getting the following:
> 
> [snip]
>
> make[1]: Entering directory `/usr/src/asterisk/apps'
> Makefile:34: *** missing separator.  Stop.
> make[1]: Leaving directory `/usr/src/asterisk/apps'
> make: *** [depend] Error 1
> 
I've not seen that error myself.  Are you trying to install Asterisk
on some weird operating system, with a non-GNU version of Make?

If so then get a copy of GNU Make and try building with that.  You may
also have to firefight other problems, so I'd suggest installing one
of the GNU/Linux distros.

-- 
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Re: [Asterisk-Users] Migrating Asterisk

2004-09-01 Thread Greg Boehnlein

On Wed, 1 Sep 2004, Jay Milk wrote:

> Hello All,
> 
> My asterisk installation has now been running for over two months
> without a hitch, and I've decided it's time to move things around a bit.
> It's currently installed on a 2.7GHz Celeron under RH9 installed on a
> 10GB "leftover" drive.  Thanks to the strange marketing method called
> "Mail-In-Rebate", I have a fresh 160GB drive ($50), and I'm itching to
> install a GenToo Linux distro.
> 
> I also have a 1.2ish GHz Duron with Mobo sitting around here, which may
> just be enough to power my (barely ever transcoding) asterisk install.
> Should be enough, even if one channel were transcoded occasionally, no?

I run Asterisk on a Pentium 133 w/ 16 megs of ram and routinely have 2-3 
ulaw to g726 transcoded sessions going on.

Works just peachy. The max I've ever done is 6 channels before I started 
having some dropouts in audio. 

So a Duron 1.2 Ghz should do just fine. ;)
 
> Let's say I start with a fresh machine, GenToo (2.4 Kernel), and a
> recent Asterisk (which one?  I'm running HEAD from 05/02/2004 right now,
> heavy on SIP, no problems), and move one of my two X100P for the timing
> source... Would it be enough to copy over the Asterisk config and VM
> files?  (yes, yes, they'll share IP addresses, so I don't have to
> reconfigure my devices)
> 
> So...
> 1) 1.2Ghz Duron, enough for transacoding a single channel?
> 2) X100P sufficient timing device for *?
> 3) Which * source does the list recommend?
> 4) \var\lib\asterisk, \etc\asterisk and zaptel.conf are all that's
> needed to migrate the current state of *?


-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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RE: [Asterisk-Users] Why are you guys promoting a Rippoff

Michael Workman [EMAIL PROTECTED] wrote:
> On your web you have a link
> 
> http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard
> 
That website is not owned by the providers of this mailing list, as
far as I know.  The WIKI is an independently-owned resource where
anyone can post anything (on topic) they like.  There's no screening
except for the fact that anyone can also remove, change or comment
on anything they don't like.

> 
> To Setup Calling with Diamondcard.us and I signed up and paid the money
> according to Stephen Karrington it was all automated... And it was
> automated to take money but when you look for service hookups or
> information you don't get it. 
> 
> I have tried now for last little while to contact them for support and
> got nothing. 
> 
Stephen Karrington is also listed on the following WIKI page, along
with a phone number and a "sales" email address:

http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants+Eastern+Europe

You could try one of those.

It's usually best to not jump to conclusions about service providers.
How long have you been waiting for the "automated" service to activate
your account?  Perhaps there's a problem with the automation at the
moment.

I have no connection with this or any other service provider, by the
way, so I can't help to speed things up.

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Re: [Asterisk-Users] Cisco 79XX SIP Ring Tones

2004-09-01 Thread Mitchell S. Sharp

On Wed, 2004-09-01 at 03:45, Shaun Ewing wrote:
> I have Asterisk installed in my home office. I have a
> residential/private number and the business numbers.
> 
> I like the private number to ring to the phone on my desk, so I use
> the ALERT_INFO to change the ring cadence.
> 
> Basically, this is what I do:
> 
> exten => xx,1,SetVar(ALERT_INFO=)
> exten => xx,2,Dial(SIP/7011|15|r)
> exten => xx,3,Voicemail,uxx
> 

This answer solved it for me.  I wasn't including the <> characters. 
It's always the small stuff!

Thanks Shaun!

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RE: [Asterisk-Users] Why are you guys promoting a Rippoff

On 1 Sep 2004 at 23:08, Michael Workman wrote:

> I would if I could contact them
> 
> The only reason I posted in here is cause I saw the Jerk "Stephen
> Karrington" posted a lot in here and he is the one runs it and told me
> to signup and its all automated...
> 
> 
> Why don't you take your problem up with them and not with this list?
> This has absolutely NOTHING to do with asterisk. Just because they're
> listed in the wiki doesn't mean that it's endorsed. 


One step better, why doesn't he go into the wiki and put a note 
regarding the company himself (that's the idea of a wiki).  

Matt Riddell
 
> > 
> > Can you tell I am upset
> > I sent emails and tried to get help from IRC and the IRC all I got
> > was no...
> > 
> > And emails All I got was
> > 
> > On your web you have a link
> > 
> > http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard
> > 
> > 
> > To Setup Calling with Diamondcard.us and I signed up and paid the
> > money according to Stephen Karrington it was all automated... And it
> > was automated to take money but when you look for service hookups or
> > information you don't get it.
> > 
> > 
> > I have tried now for last little while to contact them for support
> > and got nothing.
> > 

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RE: [Asterisk-Users] Why are you guys promoting a Rippoff

2004-09-01 Thread Michael Workman

I would if I could contact them

The only reason I posted in here is cause I saw the Jerk "Stephen
Karrington" posted a lot in here and he is the one runs it and told me to
signup and its all automated...





 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Reid A.
Forrest
Sent: Wednesday, September 01, 2004 11:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Why are you guys promoting a Rippoff

Why don't you take your problem up with them and not with this list? This
has absolutely NOTHING to do with asterisk. Just because they're listed in
the wiki doesn't mean that it's endorsed. 

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Michael 
> Workman
> Sent: Wednesday, September 01, 2004 10:10 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Why are you guys promoting a Rippoff
> 
> Can you tell I am upset
> I sent emails and tried to get help from IRC and the IRC all I got was 
> no...
> 
> And emails All I got was
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, September 01, 2004 9:30 PM
> To: [EMAIL PROTECTED]
> Subject: Re: What Happened to account
> 
> If you want to un-subscribe from a mailing list, you need to contact 
> the sender of the email directly. Or you can click the REMOVE link or 
> UNSUBSCRIBE link at the bottom of the email.
> 
> 
> 
> 
> 
> 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Michael 
> Workman
> Sent: Wednesday, September 01, 2004 9:57 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Cc: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Why are you guys promoting a Rippoff
> 
>  
> On your web you have a link
> 
> http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard
> 
> 
> To Setup Calling with Diamondcard.us and I signed up and paid the 
> money according to Stephen Karrington it was all automated... And it 
> was automated to take money but when you look for service hookups or 
> information you don't get it.
> 
> 
> I have tried now for last little while to contact them for support and 
> got nothing.
> 
> ___
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Re: [Asterisk-Users] Why are you guys promoting a Rippoff

2004-09-01 Thread William Suffill

The wiki allows everyone to post pages on whatever they wish. This
means a company can
post settings in reference to their company or anyone else could for
that matter.

On Wed, 1 Sep 2004 21:56:30 -0400, Michael Workman
<[EMAIL PROTECTED]> wrote:
> 
> On your web you have a link
> 
> http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard
> 
> To Setup Calling with Diamondcard.us and I signed up and paid the money
> according to Stephen Karrington it was all automated... And it was automated
> to take money but when you look for service hookups or information you don't
> get it.
> 
> I have tried now for last little while to contact them for support and got
> nothing.
> 
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RE: [Asterisk-Users] Why are you guys promoting a Rippoff

2004-09-01 Thread Reid A. Forrest

Why don't you take your problem up with them and not with this list? This has
absolutely NOTHING to do with asterisk. Just because they're listed in the
wiki doesn't mean that it's endorsed. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michael Workman
> Sent: Wednesday, September 01, 2004 10:10 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Why are you guys promoting a Rippoff
> 
> Can you tell I am upset
> I sent emails and tried to get help from IRC and the IRC all 
> I got was no...
> 
> And emails All I got was
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, September 01, 2004 9:30 PM
> To: [EMAIL PROTECTED]
> Subject: Re: What Happened to account
> 
> If you want to un-subscribe from a mailing list, you need to 
> contact the
> sender of the email directly. Or you can click the REMOVE link or
> UNSUBSCRIBE link at the bottom of the email.
> 
> 
> 
> 
> 
> 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Michael
> Workman
> Sent: Wednesday, September 01, 2004 9:57 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Cc: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Why are you guys promoting a Rippoff
> 
>  
> On your web you have a link
> 
> http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard
> 
> 
> To Setup Calling with Diamondcard.us and I signed up and paid 
> the money
> according to Stephen Karrington it was all automated... And 
> it was automated
> to take money but when you look for service hookups or 
> information you don't
> get it.
> 
> 
> I have tried now for last little while to contact them for 
> support and got
> nothing.
> 
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[Asterisk-Users] HFC cards and Asterisk

2004-09-01 Thread James Doherty

Hi all,

In our asterisk box we have a Fritz card and an HFC card. So far I've
had the Fritz card working well, but its time to connect the HFC card up
to the ISDN line and get it working. I've followed the instructions on
voip-info.org here:

http://www.voip-info.org/wiki-Asterisk+zaphfc+install

and here:

http://www.voip-info.org/wiki-Asterisk+zaphfc+install

But when it comes to running "ztcfg -v", I get this:

ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?

My /etc/zaptel.conf looks like this:

loadzone=nz
defaultzone=nz

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

and my /etc/asterisk/zapata.conf looks like this:

switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan=local
prilocaldialplan=local
pritrustusercid = yes

echocancel=yes
immediate=yes
group = 1
context=demo
channel => 1-2

Any ideas?
-- 
James Doherty
Zeald.com Network Operations
Ph: +64 9 415 7575, Fax: +64 9 443 9794
Web: http://www.zeald.com

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RE: [Asterisk-Users] Agents Log off

2004-09-01 Thread Joe Dennick

Put their CALLERIDNUM in the dialplan.  In my example below, all of my
extensions are 41XX, and all of the agent Ids are 43XX.  They dial 301
to log in, get prompted to enter their password, and hear the message
that they've been logged in.  To logout, they dial 302, get prompted to
enter their password, and then press '#' when prompted to enter a new
extension before hearing "Agent successfully logged out."  Yeah, there's
a bit of training, but its not too difficult.  All of my agents have
Cisco 7960 telephones where they've created memory dials (on the bottom
two buttons) for 'Queue Login' and 'Queue Logout.'

; Queue Login Extension 301
exten => 301,1,Wait(1)
exten => 301,2,AgentCallbackLogin(43${CALLERIDNUM:2}|43${CALLERIDNUM:2})
exten => 301,3,Playback(agent-loginok)
exten => 301,4,Hangup

; Queue Logout Extension 302
exten => 302,1,Wait(1)
exten => 302,2,AgentCallbackLogin(43${CALLERIDNUM:2}|'#')
exten => 302,3,Hangup

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of João Amaro
Sent: Wednesday, September 01, 2004 9:35 AM
To: Asterisk Mailing List
Subject: [Asterisk-Users] Agents Log off

Hi List,

I'm using the apllication AgentCallBackLogin so agents can login to a
queue. They just need to enter the password, the CallBack Extensions is
the ${CALLERIDNUM}

Is there a way to AgentsLogOff withou using the AgentCallBackLogin
application. I don't want the user to enter they CALLERIDNUM.

Regards

---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
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Re: [Asterisk-Users] Really Wierd softphone problem ... must read

2004-09-01 Thread Steve Kann

Actually, you're lucky that worked..
There are some soundcard drivers that are so broken, that they _always_ 
function this way; there are no "select" checkboxes under the record 
section of the windows volume controls.

We have a bunch of identical machines, and found the win2K drivers like 
this, but the winXP and Win98SE drivers were OK.

Go figure.
-SteveK
On Sep 1, 2004, at 10:07 PM, Steve Maroney wrote:
HA ! Thats was it. Thanks !!! In the past, I never understood the
configuration of the volume control. I know that sounds really newbie 
like
so I guess I never did pay enough attention  I didn't notice it.

Now that I understand what was wrong, I figured out why I changed it. I
made a change with Audacity to record the conversation of my sip 
calls. I
had to change the recording to "Stereo Mix".

Thanks guys.
As I get more experience with the telcom industry, voip, and
asterisk, i will donate more of my knowledge to this list.
On Thu, 2 Sep 2004 [EMAIL PROTECTED] wrote:
On 1 Sep 2004 at 20:00, Steve Maroney wrote:
Hey guys,
I have just developed this problem with my Windows XP box. I think it
started since I installed  XP SP2. Both SJPhone and Xlite does some
kind of bridging with the speaker out port. When ever I make a sip
call to where ever, the other party hears a lot of echoing. Well I
noticed just now when I was playing mp3's via Winamp, the music was
being played through my sip calls that I made. I am postivie that I 
do
not have microphone hooked up anywhere that is causing this. No
microphone is needed to duplicate this problem. It started with
SJPHONE. I removed and Reinstalled SJPHONE ... no luck. I install
x-lite and it does it too. I removed XP SP2, and also updated my 
sound
drivers .. no luck.  Any one ever had this heppen to them ? Please
Help.
You probably have wave out selected as well as mic for the recording
section of volume control...go into the volume control, go to
advanced, select recording, click ok, remove the checkbox from
waveout or mxier etc.  The only thing that should be selected is mic
(and maybe line in).
Matt Riddell
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RE: [Asterisk-Users] Why are you guys promoting a Rippoff

2004-09-01 Thread Michael Workman

Can you tell I am upset
I sent emails and tried to get help from IRC and the IRC all I got was no...

And emails All I got was

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 
Sent: Wednesday, September 01, 2004 9:30 PM
To: [EMAIL PROTECTED]
Subject: Re: What Happened to account

If you want to un-subscribe from a mailing list, you need to contact the
sender of the email directly. Or you can click the REMOVE link or
UNSUBSCRIBE link at the bottom of the email.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Workman
Sent: Wednesday, September 01, 2004 9:57 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Why are you guys promoting a Rippoff

On your web you have a link

http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard

To Setup Calling with Diamondcard.us and I signed up and paid the money
according to Stephen Karrington it was all automated... And it was automated
to take money but when you look for service hookups or information you don't
get it.

I have tried now for last little while to contact them for support and got
nothing.

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Re: [Asterisk-Users] Really Wierd softphone problem ... must read


HA ! Thats was it. Thanks !!! In the past, I never understood the
configuration of the volume control. I know that sounds really newbie like
so I guess I never did pay enough attention  I didn't notice it.

Now that I understand what was wrong, I figured out why I changed it. I
made a change with Audacity to record the conversation of my sip calls. I
had to change the recording to "Stereo Mix".

Thanks guys.

As I get more experience with the telcom industry, voip, and
asterisk, i will donate more of my knowledge to this list.


On Thu, 2 Sep 2004 [EMAIL PROTECTED] wrote:

> On 1 Sep 2004 at 20:00, Steve Maroney wrote:
>
> > Hey guys,
> >
> > I have just developed this problem with my Windows XP box. I think it
> > started since I installed  XP SP2. Both SJPhone and Xlite does some
> > kind of bridging with the speaker out port. When ever I make a sip
> > call to where ever, the other party hears a lot of echoing. Well I
> > noticed just now when I was playing mp3's via Winamp, the music was
> > being played through my sip calls that I made. I am postivie that I do
> > not have microphone hooked up anywhere that is causing this. No
> > microphone is needed to duplicate this problem. It started with
> > SJPHONE. I removed and Reinstalled SJPHONE ... no luck. I install
> > x-lite and it does it too. I removed XP SP2, and also updated my sound
> > drivers .. no luck.  Any one ever had this heppen to them ? Please
> > Help.
>
> You probably have wave out selected as well as mic for the recording
> section of volume control...go into the volume control, go to
> advanced, select recording, click ok, remove the checkbox from
> waveout or mxier etc.  The only thing that should be selected is mic
> (and maybe line in).
>
> Matt Riddell
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[Asterisk-Users] Why are you guys promoting a Rippoff

2004-09-01 Thread Michael Workman

 
On your web you have a link

http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard


To Setup Calling with Diamondcard.us and I signed up and paid the money
according to Stephen Karrington it was all automated... And it was automated
to take money but when you look for service hookups or information you don't
get it.


I have tried now for last little while to contact them for support and got
nothing.

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Re: [Asterisk-Users] Line death not recognized on TDM400P?

On 2 Sep 2004 at 3:00, Christian Victor wrote:
> [EMAIL PROTECTED] schrieb:
> >So basically if there is a problem with line 1 out of 12, no calls
> >are going to get through...surely this isn't expected behaviour.
> 
> What about putting all the channels in a group (1 for example) and use
> Zap/r1/number to dial. This wil use round robin to pick a different
> channel on every dial attempt.
> 
> So if there is a dead line some calls will miss but with 12 lines
> 11/12 of all calls will go through.

This is cool!  I didn't know you could do that. It's not really 
ideal, but would help in the event of failure.  Not so great if they 
unplug a line for a day (and don't want to change the configs).

I've told them if they want to unplug the line they should use normal 
grouping and unplug from 12 down, but it would be nice if the user 
could receive some type of congestion tones...

> >I spoke to a few people on IRC last night who confirmed that this is
> >the case.  It seems crazy.  If I send a fax using a fax machine it
> >works, if I dial a number with a modem it works (even a $5 modem) so
> >why can't it work in asterisk?  It used to work on the X100P's...
> >
> >I just want the card not to dial if there is no dialtone...
> >  
> 
> Maybe you could use BackgroundDetect as a workaround to check if the
> line is responding after dialing.

The problem is that dial won't give up the flow of the dialplan until 
it's finished (and with no cable plugged in, it won't give up till 
the user hangs up).

If you could specify the channel to check it would work, but alas you 
can't (i.e you could do it _before_ dial and try every line...the 
other problem is that you don't really want to be checking lines you 
know are off hook).

I don't suppose anyone knows who wrote that app?

Matt Riddell
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[Asterisk-Users] Unable to get Make

2004-09-01 Thread Norman Tomlnis

I am trying to do a make clean; make install in the asterisk source
directory and I am getting the following:


/usr/src
[EMAIL PROTECTED] src]# cd asterisk
[EMAIL PROTECTED] asterisk]# make clean; make install
for x in res channels pbx apps codecs formats agi cdr astman stdtime; do
make -C $x clean || exit 1 ; done
make[1]: Entering directory `/usr/src/asterisk/res'
rm -f *.so *.o .depend
make[1]: Leaving directory `/usr/src/asterisk/res'
make[1]: Entering directory `/usr/src/asterisk/channels'
rm -f *.so *.o .depend
rm -f busy.h ringtone.h gentone gentone-ulaw
make[1]: Leaving directory `/usr/src/asterisk/channels'
è4Bè4Bmake[1]: Entering directory `/usr/src/asterisk/pbx'
rm -f *.so *.o .depend
make[1]: Leaving directory `/usr/src/asterisk/pbx'
make[1]: Entering directory `/usr/src/asterisk/apps'
Makefile:34: *** missing separator.  Stop.
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [clean] Error 1
for x in res channels pbx apps codecs formats agi cdr astman stdtime; do
make -C $x depend || exit 1 ; done
make[1]: Entering directory `/usr/src/asterisk/res'
../mkdep -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE
-O6 -march=i686   -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\"CVS-HEAD-07/05/04-17:07:59\" -DINSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\"
-DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN
-DZAPATA_MOH  -DOPENSSL_NO_KRB5 -fPIC `ls *.c`
make[1]: Leaving directory `/usr/src/asterisk/res'
make[1]: Entering directory `/usr/src/asterisk/channels'
../mkdep -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE
-O6 -march=i686   -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\"CVS-HEAD-07/05/04-17:07:59\" -DINSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\"
-DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN
-Wno-missing-prototypes -Wno-missing-declarations   -DZAPATA_PRI
-DIAX_TRUNKING   -DCRYPTO -fPIC  `ls *.c`
make[1]: Leaving directory `/usr/src/asterisk/channels'
è4Bè4Bmake[1]: Entering directory `/usr/src/asterisk/pbx'
../mkdep -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE
-O6 -march=i686   -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\"CVS-HEAD-07/05/04-17:07:59\" -DINSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\"
-DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN  -fPIC
`ls *.c`
make[1]: Leaving directory `/usr/src/asterisk/pbx'
make[1]: Entering directory `/usr/src/asterisk/apps'
Makefile:34: *** missing separator.  Stop.
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [depend] Error 1


Any Idea?

Norm


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[Asterisk-Users] Please help to config tdm11b

2004-09-01 Thread Simon

I have a TDM400P with 1 fxo and 1 fxs module(TDM11B)
i just wanted to know how to configure the zaptel and zapata & 
extension  conf to work well
Please help me on that

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Re: [Asterisk-Users] Really Wierd softphone problem ... must read

On 1 Sep 2004 at 20:00, Steve Maroney wrote:

> Hey guys,
> 
> I have just developed this problem with my Windows XP box. I think it
> started since I installed  XP SP2. Both SJPhone and Xlite does some
> kind of bridging with the speaker out port. When ever I make a sip
> call to where ever, the other party hears a lot of echoing. Well I
> noticed just now when I was playing mp3's via Winamp, the music was
> being played through my sip calls that I made. I am postivie that I do
> not have microphone hooked up anywhere that is causing this. No
> microphone is needed to duplicate this problem. It started with
> SJPHONE. I removed and Reinstalled SJPHONE ... no luck. I install
> x-lite and it does it too. I removed XP SP2, and also updated my sound
> drivers .. no luck.  Any one ever had this heppen to them ? Please
> Help.

You probably have wave out selected as well as mic for the recording 
section of volume control...go into the volume control, go to 
advanced, select recording, click ok, remove the checkbox from 
waveout or mxier etc.  The only thing that should be selected is mic 
(and maybe line in).

Matt Riddell
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Re: [Asterisk-Users] Line death not recognized on TDM400P?

On 1 Sep 2004 at 9:00, Andrew Kohlsmith wrote:

> On Wednesday 01 September 2004 01:15, [EMAIL PROTECTED] wrote:
> > They would like to be able to unplug lines and use them for other >
> purposes at times.
> 
> Out of curiosity, why are they unplugging the lines?  i.e. what are
> these "other purposes" ?
> 
Well number 1 was to check what would happen if there was a line 
failure.

But the other purposes were to bring online lots of fax machines in 
time of need?!

Cheers,

Matt
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[Asterisk-Users] Really Wierd softphone problem ... must read



Hey guys,

I have just developed this problem with my Windows XP box. I think it
started since I installed  XP SP2. Both SJPhone and Xlite does some
kind of bridging with the speaker out port. When ever I make a sip call to
where ever, the other party hears a lot of echoing. Well I noticed just
now when I was playing mp3's via Winamp, the music was being played
through my sip calls that I made. I am postivie that I do not have microphone
hooked up anywhere that is causing this. No microphone is needed to
duplicate this problem. It started with SJPHONE. I removed and Reinstalled
SJPHONE ... no luck. I install x-lite and it does it too. I removed XP
SP2, and also updated my sound drivers .. no luck.  Any one ever had this
heppen to them ? Please Help.

I also tried other sip clinets on another pc with no problem so it doesn't
seem to be the server.

Thank you,
Steve Maroney

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Re: [Asterisk-Users] Line death not recognized on TDM400P?

2004-09-01 Thread Christian Victor

[EMAIL PROTECTED] schrieb:
So basically if there is a problem with line 1 out of 12, no calls 
are going to get through...surely this isn't expected behaviour.
 

What about putting all the channels in a group (1 for example) and use 
Zap/r1/number to dial. This wil use round robin to pick a different 
channel on every dial attempt.

So if there is a dead line some calls will miss but with 12 lines 11/12 
of all calls will go through.

I spoke to a few people on IRC last night who confirmed that this is 
the case.  It seems crazy.  If I send a fax using a fax machine it 
works, if I dial a number with a modem it works (even a $5 modem) so 
why can't it work in asterisk?  It used to work on the X100P's...

I just want the card not to dial if there is no dialtone...
 

Maybe you could use BackgroundDetect as a workaround to check if the 
line is responding after dialing.

Christian
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[Asterisk-Users] Newbie: Asterisk Config to Replace Lucent Partner

2004-09-01 Thread Staalenburg, Juan

Newbie to Asterisk (but seasoned pro at networking/linux/etc).  Looking to
replace a Lucent Partner Phone system with an Asterisk based solution.

3 Incoming Analog Phone Lines
10 Phone Extensions

Need:

Auto-Attendant
Voicemail
Caller ID
Transfer
Conferencing

Outgoing calls will be made through the 3 analog lines (not internet).

Any information or some help in pointing me in the right direction for a
sample configuration would be appreciated.


Regards,
___
Juan Staalenburg


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[Asterisk-Users] Festival TTS & mbrola ?

2004-09-01 Thread Danny Zak

Hello Asterisk,

  noticed that the mbrola really adds a extra dimension to tts; anyone
  got any experience with running this together with the festival and
  * ?

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Re: [Asterisk-Users] Faxing with an IAXy

2004-09-01 Thread Scott Petersen

On Mon, Aug 30, 2004 at 06:57:34PM +0100, Ben Merrills wrote:
> Hi,
> 
>  
> 
> Has anyone got faxing working via an IAXy? I'm having trouble getting it
> working with a fax machine here. The IAXy works fine, I can make
> incoming/outgoing calls via it, however our PDQ takes a few redials to
> get data over to the dialled party, and faxing doesn't seem to work at
> all.
> 
>  
> 
> I have turned off all the echo cancellation stuff, just in case that was
> an issue. Anything I should try, has anyone had any luck doing this
> either here in the UK or elsewhere?
> 

I spent many, many days attempting the same thing. I could get 5-6 pages sent 
relatively reliably but slowly. Then about the 7th page it would start retrying pages 
and retrain slower and slower until it crapped out between the 8th and 17th page. This 
would average > 90sec per page. When I took the IAXy and * out of the loop it would 
take approximately 10-15 seconds per page. This is with a 100mb network and the * box 
and IAXy plugged into the same switch. The Zap interface had all echo cancellation 
turned off etc. etc. I tried all sorts fo different configurations that left the only 
common denominator being the IAXy.

 From what I have read, it is an issue with timing. That perhaps since the IAXy does 
not have a similiar timer to the * box there is enough drift that a  few packets get 
lost when crossing a timing boundary. I don't completely understand the fax 
methodology but I do know that it is very timing dependant and does not work properly 
with the IAXy. This is with firmware 17 and 18. Maybe a later firmware can resolve it, 
who knows.

Cheers
Scott Petersen
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RE: [Asterisk-Users] Asterisk SIP between two networks

2004-09-01 Thread Sergio Serrano

Hi all,
I'm desperate,
if I put bindaddr=192.168.20.10, I obtain the next:

Sip read: 
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK2174f136
From: ;tag=as05db6abc
To:
;tag=84448f3c7053227cca70775302748de3.a866

Call-ID: [EMAIL PROTECTED]
CSeq: 102 REGISTER

WWW-Authenticate: Digest realm="voztele.com",
nonce="41365c4cf9c69cc73a429f27813652ded65fc483"
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0

But If i put bindaddr=0.0.0.0, I obtain yhe next:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK54679e05
From: ;tag=as294baf04
To:
;tag=84448f3c7053227cca70775302748de3.e5c8

Call-ID: [EMAIL PROTECTED]: 103 REGISTER

WWW-Authenticate: Digest realm="voztele.com",
nonce="41365cfc1947f24b5cd03bb5bca062540243dc39"
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0

It is a bug? Why if I put bindaddr=0.0.0.0 packet received by asterisk
is broken?

Could anyone help me?

Regards,
srsergio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano
Enviado el: jueves, 02 de septiembre de 2004 0:28
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks


I just use localnet parameter in next way:
localnet=192.168.20.0/255.255.255.0
localnet=172.28.240.0/255.255.240.0

Any idea more?

Regards,
srsergio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Kevin
Walsh Enviado el: miércoles, 01 de septiembre de 2004 19:16
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks


Sergio Serrano [EMAIL PROTECTED] wrote:
> SIP Provider<--->ADSL router<---localnet
> 192.168.20.0--->ASTERISK<---localnet 172.24.240.0--->softphones
> 
> first localnet 192.168.20.0
> second localnet 172.28.240.0
> in second localnet we have softphone and the first localnet is
> connected to ADSL router to connect to our SIP provider.
> 
> if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't
> register in my SIP provider. If I put 192.168.20.10 in bindaddr I can 
> register in my SIP provider but softphones can't register into 
> asterisk. I 'm using asterisk RC1.
> 
You probably need to use the "localnet" setting in sip.conf.  See here
for more sip.conf-related information:

http://www.voip-info.org/wiki-Asterisk+config+sip.conf

On the other hand, you could use an IAX2 provider and side-step the
issue altogether.

-- 
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  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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RE: [Asterisk-Users] All you polycom folks.....

2004-09-01 Thread John Baker

Sip 1.3.1 is the latest.

John

On Wed, 2004-09-01 at 07:56, Matthew Marlowe wrote:
> I left my phone at home I think Im using sip 1.3.1.. It's 1.3.
> something
> 
> Asterisk CVS-HEAD-05/12/04-13:23:20, Copyright (C) 1999-2004 Digium. 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Brent
> Franks
> Sent: Tuesday, August 31, 2004 10:14 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] All you polycom folks.
> 
> Just out of curiosity,
> 
> What version of CVS and Polycom SIP software are you running happily?
> 
> Are you running SIP 2.3.0 yet?  2.2.0?  2.1.0?
> 
> I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1
> with poor results.  Transferring did not work as expected.  Using the #
> key to do blind transfers after a call was on hold did not work.
> 
> Just curious.
> 
> Thanks,
> 
> - Brent
> 
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> 


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[Asterisk-Users] Analog -> ip sip softphone on Fritz Capi - strong reverb ?

2004-09-01 Thread Robert Rozman

Hi,

I have problems that seems to be common when communicate from analog phone
to digital (ISDN). I have ISDN passive card Fritz and would kindly ask for
following info:

- are there any settings that can limit reverb ?
- are there any more info about capi.conf parameters and how to use in
Asterisk and setup CAPI device properly ?

Thanks in advance,

Robert.


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[Asterisk-Users] zaphfc crashes Linux

2004-09-01 Thread Leo Ann Boon

Hi all,
I'm having serious problem getting zaphfc to work on my box. I d/l'd 
bri-stuff-0.1.0RC3/RC4a and followed the instructions to the dot. 
Everything builds fine. But, when 'make load' the whole machine will 
freeze. Anyone had the same problem and managed to solve it? I'm using a 
Billion HFC PCI card on Trustix 2.0 running kernel 2.4.26.

As a side note, I feel that the bri-stuff is too brittle. For starters, 
it doesn't work with * 1.0RC2 and I've a feeling it won't work with 1.0 
final as well. Perhaps the community has to coerce kapejod to get 
bri-stuff work with mainstream * versions? Just my $0.02.

Cheers and TIA.
Leo
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RE: [Asterisk-Users] Asterisk SIP between two networks

2004-09-01 Thread Sergio Serrano

I just use localnet parameter in next way:
localnet=192.168.20.0/255.255.255.0
localnet=172.28.240.0/255.255.240.0

Any idea more?

Regards,
srsergio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Kevin
Walsh
Enviado el: miércoles, 01 de septiembre de 2004 19:16
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks


Sergio Serrano [EMAIL PROTECTED] wrote:
> SIP Provider<--->ADSL router<---localnet 
> 192.168.20.0--->ASTERISK<---localnet 172.24.240.0--->softphones
> 
> first localnet 192.168.20.0
> second localnet 172.28.240.0
> in second localnet we have softphone and the first localnet is 
> connected to ADSL router to connect to our SIP provider.
> 
> if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't 
> register in my SIP provider. If I put 192.168.20.10 in bindaddr I can 
> register in my SIP provider but softphones can't register into 
> asterisk. I 'm using asterisk RC1.
> 
You probably need to use the "localnet" setting in sip.conf.  See here
for more sip.conf-related information:

http://www.voip-info.org/wiki-Asterisk+config+sip.conf

On the other hand, you could use an IAX2 provider and side-step the
issue altogether.

-- 
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  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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[Asterisk-Users] h323 - forcing user authentication

2004-09-01 Thread Marcin Mazurek

Hi,

is there a way to force a user authentication using h323 channel from
asterisk sources? Do I have to use gatekeeper for this?

Is there any way to do it in h323.conf just like in sip.conf?

eg:
[mazek]
secret=xx
auth=md5


tia

mazek

-- 
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[Asterisk-Users] X100P + Call-Waiting - Flash how-to.

2004-09-01 Thread Guillaume Giraudon

Hi all

I'm pretty sure someone must have done this before but I couldnt find any
trace of it on the web so I thought I would drop a note about how I ended up
doing it. I have also posted this info on voip-info.

Warning : This is not very elegant and I'm currently trying to write a patch
in order to make it better but so far, this the only way I've gotten this to
work.

Scenario :
I have an asterisk box with an X100P card. When my phone line rings, it ring
my SIP Phone (a Cisco 7940).
I've got a call-waiting feature on my line and couldnt figure out how to
trigger a flash in order to go from one call to another.

Solution :

1st - The inbound context (in extensions.conf of course)
[pstninbound]
exten=>s,1,Dial(SIP/cisco7940,40|Tt)
exten=>s,2,Voicemail(u1000)
exten=>s,3,Voicemail(b1000)

The Tt option will allow you to transfer by hitting the # key.

2nd - The flash extension

Now, somewhere in your extensions file, create a context that similar to
this :

exten=>604,1,Flash()
exten=>604,2,Dial(SIP/cisco7940)


By transfering a ZAP call to that extension, the line is flashed before
ringing back to you.

There are 2 ways to use this setup : Either by using the # key on an inbound
call, or by using the BlndXfr key (if you do so, you can actually take the
Tt our of the Dial sequence in your inbound context).

When hearing the call-waiting tones, just blind transfer the call to
extension 604. You ZAP channel will be flashed and it will ring back to you.

Hope this helps.

G.

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RE: [Asterisk-Users] Migrating Asterisk

2004-09-01 Thread Jay Milk

Thanks for the quick response -- I should have clarified this a little
more.

I was using this board before, but couldn't get ztdummy to work because
the board had the wrong USB controller on it.  I switched boards and
then added the X100P.  I would rely solely on the X100P for timing if I
go back to this board.  This is a home-pbx, so I only need timing for VM
really -- and maybe someday I'll look into getting meetme to work.

> -Original Message-
> From: William Suffill [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, September 01, 2004 3:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Migrating Asterisk
> 
> 
> 2) Depending how much timing you need to do X100P or ztdummy 
> could even work just fine.

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[Asterisk-Users] TDM40B hangup on fax or data modem carrier

2004-09-01 Thread Arnaud Pignard

Hi !
I have a TDM40B and i try to use it connected to modem for incoming call 
data transfert.

I have no problem to use it with a phone and a talk communication work fine.
But when we try to use with modem, with most modem, we got data carrier for 
few seconds and channel hungup.

< [ TYPE: Null Frame (4) SUBCLASS: N/A (3) ] [Zap/4-1]
-- Zap/4-1 is ringing
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/4-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/4-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/4-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/4-1]
<< [ TYPE: Null Frame (4) SUBCLASS: N/A (4) ] [Zap/4-1]
-- Zap/4-1 answered SIP/213.161.193.64-08142788
<< [ HANGUP (NULL) ] [Zap/4-1]
-- Hungup 'Zap/4-1'
I try to configure channel in different mode, without echo cancellation and 
seems same problem.

This configuration is working perfectly with HandyTone from GrandStream.
Our zapatel.conf look like this (but i have also test with light 
configuration) :

signalling=fxo_ls   ; try also ks ...
group=1
relaxdtmf=yes   ; make no difference
context=sip
echocancel=no
faxdetect=no; make no difference
channel => 1-4
All call are incoming call (from PSTN or SIP G711 - ASTERISK - TDM40B - MODEM)
Modem said "no carrier" when answering with ATA.
With a fax machine, i think we will get same problem haven't yet test)
Any idea ?
Thanks for help !
--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
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Re: [Asterisk-Users] Line death not recognized on TDM400P?

On 1 Sep 2004 at 7:30, Rich Adamson wrote:

> > A customer of mine has 3 TDM400P cards in a box running asterisk. 
> > On each card he has four FXO modules.  
> > 
> > I have set up the dialplan to dial via group 1 for an outgoing call.
> > 
> > Channels 1-12 are in group 1.
> > 
> > If he plugs a telephone cable into socket 2 or 3 etc, but not 1,
> > when he dials out, it still tries to make the call via socket 1.
> > 
> > Straight away the console says that it has dialed the number via g1
> > and that it is connecting sip/bla with zap/1-1 (or some such)...
> > 
> > On my X100P I get a red alarm if the phone cable is not plugged in. 
> > Is there any way to do this with the TDM400P?
> > 
> > They would like to be able to unplug lines and use them for other
> > purposes at times.
> > 
> > Make sense?
> > 
> > I kinda thought that asterisk would realise that nothing was 
> > connected to the TDM card and try the second socket, the third
> > etc...
> 
> It makes sense, but the code to detect unused rj11's is not in * now.
> 
> In fact, you'll find that unplugging and replugging the rj11's will
> cause * to fail after a while. (At least that was the case about a
> month ago and there really haven't been any changes to the fxo
> software for some time.)
> 
> There are no alarms or other indicators available that would suggest a
> port has failed or is unavailable.
> 
So basically if there is a problem with line 1 out of 12, no calls 
are going to get through...surely this isn't expected behaviour.

Is there any way to fix this (I'm the New Zealand distributor of 
Digium products and if this is the case, they will be returning to 
their propreity pabx)?

Should I open a bugnote?

I spoke to a few people on IRC last night who confirmed that this is 
the case.  It seems crazy.  If I send a fax using a fax machine it 
works, if I dial a number with a modem it works (even a $5 modem) so 
why can't it work in asterisk?  It used to work on the X100P's...

I just want the card not to dial if there is no dialtone...

Matt
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RE: [Asterisk-Users] UK Disconnect supervision with TDM400P

2004-09-01 Thread Edward Eastman

Hi, thanks for the reply, only just got round to having a look at it again
(annoying how real life gets in the way of the important stuff ;) 

I've had a go at ramping up the tx/rx gain but it doesn't seem to make any
difference.  FWIW it's the same with the module in normal fcc mode.

Does anyone know if bt do normally provide disconnect supervision or whether
it has to be done with e.g. busydetect (and can either be detected by the
tdm400p in uk mode)?

Thanks

Ed

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard Scobie
Sent: 28 August 2004 21:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UK Disconnect supervision with TDM400P



Edward Eastman wrote:

> 
> I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN
line,
> loading wcfxs with OPERMODE=UK.  All's working well, except if I get an
> incoming call through my bt line, and the remote party hangs up, I get
> approx 20secs of the bt line hungup tone before asterisk hangs up, which
> leads (if nothing else) to the well documented 20secs of beep on vm
problem
> :)
> 
> I have tried: busydetect=yes / busycount=7 / other busycounts /
> callprogess=yes but none of these make any difference.  I have
> loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks
> signalling.
> 

Try increasing your RX gain in 1db steps, until it reliably hangs up.

I had a box with X100Ps which busydetected perfectly with default gain 
settings. When they were replaced with TDM FXOs, busydetect stopped 
working and I needed 3db of RX gain added to get it working again.

Regards,

Richard
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Re: [Asterisk-Users] SMS & Asterisk - an explanation

2004-09-01 Thread Axel Eble

On Wed, 1 Sep 2004 16:38:30 +0100, Asterisk <[EMAIL PROTECTED]> wrote:
> Oh great. BT at it's best.
> 
> Spoken to 4 different product managers / isdn help desk / customer service
> weenies. Each one says "Eh? What?" and I've got to point them to your link.
> "I'll get back to you on that" is the next response.
> 
> Still waiting 

Hah. Am I happy to see that BT isn't any better than Deutsche Telekom.

Axel

-- 
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VoIP: [EMAIL PROTECTED] * cell: +49.178.285-3265
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Re: [Asterisk-Users] Dynamic dialplan

2004-09-01 Thread Chris A. Icide

On 06:25 AM 9/1/2004, Juan Jose Comellas wrote:
>We intend to use Asterisk with a very large dialplan (with a lot of
>functionality for 3000+ users). Each user will be able to change several of
>his parameters in the dialplan, so we will be forced to reload the diaplan
>constantly. Has anybody else any previous experience with a similar
>installation? There are some things that we'd like to know, if anybody can
>help us. These are:
>
>- Is this something that can be done safely with Asterisk?
Yes, see the wiki on scaling issues and hardware requirements
>
>- Can we have a diaplan configuration update every 5 or 10 minutes without
>service interruption?
>
A better idea would be to implement a static dialplan that access a database...
for example...
[inbound-pri]
; this context handles all inbound calls arriving on our PRI
; dnid is 4 digits from the telco
exten => 3748,1,DBGet(user-chan=inbound/${EXTEN})
exten => 3748,2,Dial(${user-chan})
just a very simple example of the concept
>- What happens to new calls while the dialplan configuration is being
>reloaded?
Guess this depends upon where in the process of loading a huge extensions 
configuration is when the call comes in...

>
>- What happens to active calls after the dialplan configuration is updated?
Active channels will remain active.   In other words a call between to 
channels will remain in place during a reload.

>
>- Can we do partial updates of the dialplan (e.g. update a specific context
>instead of the whole dialplan configuration)?
I believe the only way to force a dialplan reload is through the CLI 
command: extensions reload, which forces extensions.conf to be loaded in full

>
>- Can Asterisk have its dialplan in a database instead of having it 
always in
>memory?

Yep, see above comment for just one of the methods to do this, there are 
quite a few.

>
>
>Thanks.
>
>--
>Juan Jose Comellas
>([EMAIL PROTECTED])
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Re: [Asterisk-Users] Migrating Asterisk

2004-09-01 Thread William Suffill

1) should be more than enuf for 1 channel. I use a P2 400 here for
testing and it worked ok for transcoding besides the schedule notices.

2) Depending how much timing you need to do X100P or ztdummy could
even work just fine.

3. -head 

4. i'd rebuild it from src and just copy your configs and any of your stored VM

On Wed, 1 Sep 2004 14:39:12 -0500, Jay Milk <[EMAIL PROTECTED]> wrote:
> Hello All,
> 
> My asterisk installation has now been running for over two months
> without a hitch, and I've decided it's time to move things around a bit.
> It's currently installed on a 2.7GHz Celeron under RH9 installed on a
> 10GB "leftover" drive.  Thanks to the strange marketing method called
> "Mail-In-Rebate", I have a fresh 160GB drive ($50), and I'm itching to
> install a GenToo Linux distro.
> 
> I also have a 1.2ish GHz Duron with Mobo sitting around here, which may
> just be enough to power my (barely ever transcoding) asterisk install.
> Should be enough, even if one channel were transcoded occasionally, no?
> 
> Let's say I start with a fresh machine, GenToo (2.4 Kernel), and a
> recent Asterisk (which one?  I'm running HEAD from 05/02/2004 right now,
> heavy on SIP, no problems), and move one of my two X100P for the timing
> source... Would it be enough to copy over the Asterisk config and VM
> files?  (yes, yes, they'll share IP addresses, so I don't have to
> reconfigure my devices)
> 
> So...
> 1) 1.2Ghz Duron, enough for transacoding a single channel?
> 2) X100P sufficient timing device for *?
> 3) Which * source does the list recommend?
> 4) \var\lib\asterisk, \etc\asterisk and zaptel.conf are all that's
> needed to migrate the current state of *?
> 
> TIA!
> 
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Re: [Asterisk-Users] limit the length of extensions

2004-09-01 Thread Chris A. Icide

On 07:05 AM 8/31/2004, Deon Rodden wrote:
>How do I limit the length of an extension? In my test IVR/Automated
>Attendant (whatever it's called), at the beginning it plays "if you know
>your parties 3 digit extension, you may enter it now) and then it gives
>a list of options. If the caller puts the 3 digit extension, it goes
>through fine, if they press 1, or 2 it goes to the selected menu option,
>but if they dial 91235551212 it dials that phone number. Which of
>course, is a big security risk.
>
>Is there a way to limit the length of an extension for an incoming call?
>My only solution right now is to duplicate ever single extension (about
>50 of them) in a seperate context, one that does not have the _9.
>extension in it, and then make the call in menu have access to that
>context.  However, if I put a limit in the entire context of 3 digits,
>then my coworkers who's phones are in that context can only dial each
>other, not 9 and an outside number. So it has to be an incoming limit or
>something.
It sounds like you need to break your dialplan into more focused 
contexts.  Create a context for access to an outside line (you can even 
break this down into access for toll-free, long distance, local, 
local-toll, international, '976' numbers, etc.  Create contexts for each 
set of company extensions, create contexts for all the ivr systems 
separately, so only the options you want are available.  Then use include 
statements to only include the access you want for each context.  When I 
set systems up like this I also include 'anti' contexts.  For example, if I 
include a local context that allows 7, 10, or 11 digit dialing, I also 
include a context !local that plays back an unauthorized message if someone 
dials a matching number.  This way you can not only catch access to 
unauthorized features by implicitly denying access, but you provide the 
user with a reason why they are not able to place the call they just dialed.

-Chris
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[Asterisk-Users] Migrating Asterisk

2004-09-01 Thread Jay Milk

Hello All,

My asterisk installation has now been running for over two months
without a hitch, and I've decided it's time to move things around a bit.
It's currently installed on a 2.7GHz Celeron under RH9 installed on a
10GB "leftover" drive.  Thanks to the strange marketing method called
"Mail-In-Rebate", I have a fresh 160GB drive ($50), and I'm itching to
install a GenToo Linux distro.

I also have a 1.2ish GHz Duron with Mobo sitting around here, which may
just be enough to power my (barely ever transcoding) asterisk install.
Should be enough, even if one channel were transcoded occasionally, no?

Let's say I start with a fresh machine, GenToo (2.4 Kernel), and a
recent Asterisk (which one?  I'm running HEAD from 05/02/2004 right now,
heavy on SIP, no problems), and move one of my two X100P for the timing
source... Would it be enough to copy over the Asterisk config and VM
files?  (yes, yes, they'll share IP addresses, so I don't have to
reconfigure my devices)

So...
1) 1.2Ghz Duron, enough for transacoding a single channel?
2) X100P sufficient timing device for *?
3) Which * source does the list recommend?
4) \var\lib\asterisk, \etc\asterisk and zaptel.conf are all that's
needed to migrate the current state of *?

TIA!

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[Asterisk-Users] Asterisk, newbie, fwd and is this jitter?

2004-09-01 Thread Quentin Cope

Hi

I've been playing around with asterisk for a while now at home, just trying
to understand a bit of the technology and seeing what I could get up and
running. Here's where I am at:

I bought myself an X100P card and got an asterisk server up and running on a
gentoo linux distro. I got two windows PC clients, running x-lite to work
with this and got basic functionality running. After some help from this
mailing list I was able to add festival and music on hold to asterisk and
this all seems ok.

I then looked at connecting asterisk to FWD. First I set it all up as a
regular SIP connection and then I moved it onto using IAX. However no matter
how I connect to FWD, festival and music on hold sound awful but fine over
the analogue connection.

If anyone has FWD running, is what you hear on 290718 as good as it gets? Is
this "jitter" that appears in posts to the mailing list?

Regards

Quentin

BTW, I've disconnected the analogue just in case my dial plan is not too
clever ;-)

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[Asterisk-Users] Using an analog modem through asterisk (zap channels)

2004-09-01 Thread Rob Fugina

I've tried this before, with no luck.  I've got to try again this
evening, and I'm looking for some help.

Here's my configuration -- pretty simple, really.

Asterisk box -> T100P -> TA750(20FXS/4FXO) -> phones and outgoing lines

I have an analog modem (Ok, it's a TIVO) that I need to be able to
dial out.  Right now, I have the modem connected directly to an
outgoing line that asterisk is also connected to.  As long as the
line's not in use, the modem can pick it up and use it.

What I NEED to be able to do, is connect the modem to one of the
TA750's FXS ports, and dial out.  And it's not the dial plan that's
the problem.  I've gone so far as to set up an FXS port to
automatically Dial() one of the outside lines, which gives the modem
the outside dialtone immediately.

The problem has been that the modem just can't negotiate a connection
with the remote end.

I've tried turning off echo cancelation on the FXS port, and I
probably tried the same thing on the FXO port (which isn't a good
option, since voice calls need to go out over that port, too).

Is there something special I should be doing to make this work?  I've
never seen an answer either way...

Thanks,
Rob
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Re: [Asterisk-Users] Help Me - SIP Phones ( No Voice) !!!!

Also make sure there isn't any packet filtering enabled on the BSD box as
well.

Thank you,
Steve Maroney

On Wed, 1 Sep 2004, Jefferson Carvalho wrote:

> Hello list,
>
> I've posted my problem on BSD list and i still have the
> problem.
> The remote side receives the call , but there's no voice
> on the call.
> I tried everything about possible NAT problems ..
> but ther're on same net.
>
> My platform:
>
> FreeBSD 5.2.1-Release
> Asterisk 1.0-RC2
> soft phones : X-Lite
>
>  
> -- Executing Dial("SIP/1260-a7ae", "SIP/1262|20") in new stack
> -- Called 1262
> -- SIP/1262-c597 is ringing
> -- SIP/1262-c597 answered SIP/1260-a7ae
> -- Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
> - Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
> Sep  1 14:53:17 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum
> retries exceeded on call
> [EMAIL PROTECTED] for seqno 11288
> (Non-critical Response)
> *
>  > My sip.conf
>
> *[1260]
> type=friend
> username=1260
> secret=jeff
> context=sip
> qualify=300
> mailbox=1260
> callerid="Jefferson Carvalho" <1260>
> host=dynamic
> nat=no
> canreinvite=no
> allow=gsm
> ;
> [1262]
> type=friend
> context=sip
> username=1262
> secret=1262
> qualify=300
> callerid="Ialle" <1262>
> host=dynamic
> nat=no
> canreinvite=no
> allow=gsm
> ;
>
> *>> My extensions.conf
> *
> [general]
>
> static=yes
> writeprotect=no
>
> [globals]
>
> CONSOLE => Console/dsp
> IAXINFO => guest
> TRUNK => Zap/g2
> TRUNKMSD => 1
>
> [sip]
>
> exten => 1260,1,Dial(SIP/1260,20)
> exten => 1261,1,Dial(SIP/1261,20)
> exten => 1262,1,Dial(SIP/1262,20)
>
> Best Regards,
>
> -Jefferson Carvalho
>  IT Analist
>  Credishop S/A
>  Teresina-PI-Brasil
>  5586-94321901
>
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Re: [Asterisk-Users] Help Me - SIP Phones ( No Voice) !!!!

Are these softphones ? If so, make sure there isn't any packet filtering
(firewall) taking place. I dont have too much experience with Asterisk
& VOIP so the next thing I would try is not registering with
the server and make call directly to the other phone via its IP.

Hope this helps.

Thank you,
Steve Maroney

On Wed, 1 Sep 2004, Jefferson Carvalho wrote:

> Hello list,
>
> I've posted my problem on BSD list and i still have the
> problem.
> The remote side receives the call , but there's no voice
> on the call.
> I tried everything about possible NAT problems ..
> but ther're on same net.
>
> My platform:
>
> FreeBSD 5.2.1-Release
> Asterisk 1.0-RC2
> soft phones : X-Lite
>
>  
> -- Executing Dial("SIP/1260-a7ae", "SIP/1262|20") in new stack
> -- Called 1262
> -- SIP/1262-c597 is ringing
> -- SIP/1262-c597 answered SIP/1260-a7ae
> -- Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
> - Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
> Sep  1 14:53:17 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum
> retries exceeded on call
> [EMAIL PROTECTED] for seqno 11288
> (Non-critical Response)
> *
>  > My sip.conf
>
> *[1260]
> type=friend
> username=1260
> secret=jeff
> context=sip
> qualify=300
> mailbox=1260
> callerid="Jefferson Carvalho" <1260>
> host=dynamic
> nat=no
> canreinvite=no
> allow=gsm
> ;
> [1262]
> type=friend
> context=sip
> username=1262
> secret=1262
> qualify=300
> callerid="Ialle" <1262>
> host=dynamic
> nat=no
> canreinvite=no
> allow=gsm
> ;
>
> *>> My extensions.conf
> *
> [general]
>
> static=yes
> writeprotect=no
>
> [globals]
>
> CONSOLE => Console/dsp
> IAXINFO => guest
> TRUNK => Zap/g2
> TRUNKMSD => 1
>
> [sip]
>
> exten => 1260,1,Dial(SIP/1260,20)
> exten => 1261,1,Dial(SIP/1261,20)
> exten => 1262,1,Dial(SIP/1262,20)
>
> Best Regards,
>
> -Jefferson Carvalho
>  IT Analist
>  Credishop S/A
>  Teresina-PI-Brasil
>  5586-94321901
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] FXO Disconnect supervision

2004-09-01 Thread Scott Laird

On Sep 1, 2004, at 11:35 AM, Eric Wieling wrote:
On Wed, 2004-09-01 at 12:26, Glen Johnson wrote:
Does anyone have disconnect supervision working on their TDM400P or 
X101P
cards (to/from the telco)?
Yes on all 4 my X100P/TDM400PFXO ports across 3 servers in two states.
It Just Works.  Of course I'm in the USA.
This brings up an interesting point--disconnect supervision *mostly* 
works for me with a X100P in the US.  The exception is when calls go to 
voicemail; I frequently end up with ~90 seconds of dialtone instead of 
a message or a clean disconnect.  This has remained constant for 6 
months, up through RC1.

Any suggestions?
Scott
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RE: [Asterisk-Users] Group Dial

2004-09-01 Thread Robinson Tim-W10277

Title: Message



What 
is your definition of TRUNKBP ?
 
It is 
probably because that channel is being answered first
 
Rgds
Tim

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Tomica 
  CrnekSent: 01 September 2004 15:19To: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] Group 
  Dial
  Hi 
  everyone,
   
  I want to have a 
  group and dial multiple phones/lines simultaneously. If I use this Dial 
  command:
   
  exten => 
  222,2,Dial(${TRUNKBP}/246&SIP/258&${TRUNKBP}/243,20,tTr)
   
  ... all phones 
  ring just once, after that only the first one continues ringing and only that 
  one can answer. Can anyone tell me why?
   
  thanks!
  Tomica
   
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Re: [Asterisk-Users] latest CVS build won't load

Hmmm... Well what that means is that the code is using pthread_create()
instead of ast_pthread_create(), it's not a major thing, all you would have
to do is go through all the affected modules and replace pthread_create with
ast_pthread_create, but this should probably be fixed in CVS too!

-Chris

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Re: [Asterisk-Users] latest CVS build won't load

2004-09-01 Thread Alok K. Dhir

I did...
Chris Shaw wrote:
Make sure you delete your /usr/lib/asterisk directory before installing a
new CVS copy...
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--
Alok K. Dhir <[EMAIL PROTECTED]>
Symplicity Corporation
http://solutions.symplicity.com
703 351 6987 (w) | 703 351-6357 (f)
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Re: [Asterisk-Users] latest CVS build won't load

Make sure you delete your /usr/lib/asterisk directory before installing a
new CVS copy...

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[Asterisk-Users] latest CVS build won't load

2004-09-01 Thread Alok K. Dhir

Fails on loading several of the chan_*.so modules with "undefined symbol 
__use_ast_pthread_create_instead__". 

Notably, these same modules complain during compilation "implicit 
declaration of function __use_ast_pthread_create_instead__".

Ideas?
--
Alok K. Dhir <[EMAIL PROTECTED]>
Symplicity Corporation
http://solutions.symplicity.com
703 351 6987 (w) | 703 351-6357 (f)
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[Asterisk-Users] Broken sound in VoiceMail

2004-09-01 Thread Ben Merrills









It seems voicemail recordings have broken sound. It
cuts out randomly throughout the recording. Has anyone had any similar experiences?

 

I’ve included some snips of my voicemail.conf

 

Cheers,

 

Ben

 

--SNIP---

[general]

; Default formats for writing Voicemail

;format=g723sf|wav49|wav

format=wav

; Who the e-mail notification should appear to come
from

[EMAIL PROTECTED]

;[EMAIL PROTECTED]

; Should the email contain the voicemail as an
attachment

attach=yes

; Maximum length of a voicemail message in seconds

;maxmessage=180

; Minimum length of a voicemail message in seconds

;minmessage=3

; Maximum length of greetings in seconds

;maxgreet=60

; How many miliseconds to skip forward/back when
rew/ff in message playback

skipms=3000

; How many seconds of silence before we end the
recording

maxsilence=10

; Silence threshold (what we consider silence, the
lower, the more sensitive)

silencethreshold=100

; Max number of failed login attempts

maxlogins=3

; If you need to have an external program, i.e.
/usr/bin/myapp

; called when a voicemail is left, delivered, or your
voicemailbox 

; changes, uncomment this:

;externnotify=/usr/bin/myapp

; For the directory, you can override the intro file
if you want

;directoryintro=dir-intro






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Re: [Asterisk-Users] FXO Disconnect supervision

2004-09-01 Thread Eric Wieling

On Wed, 2004-09-01 at 12:26, Glen Johnson wrote:
> Does anyone have disconnect supervision working on their TDM400P or X101P 
> cards (to/from the telco)?

Yes on all 4 my X100P/TDM400PFXO ports across 3 servers in two states. 
It Just Works.  Of course I'm in the USA.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] FXO Disconnect supervision

2004-09-01 Thread Glen Johnson

Does anyone have disconnect supervision working on their TDM400P or X101P 
cards (to/from the telco)?

Googling "telco disconnect supervision," I found the following description:
http://mirror.lcs.mit.edu/telecom-archives/telecom-archives/TELECOM_Digest_Online/1559.html
[cut]
Re: How Do I Get "Kewlstart" From my Phone Company?
Paul A Lee
Tue, 24 Aug 2004 16:58:16 -0400
...
It appears that "kewlstart" is just a coined name for loop signaling
with disconnect supervision. Disconnect supervision is also called
"calling party control", "forward disconnect", "open loop disconnect",
"open switch interval", "adjunct control", and perhaps other
names. What is supposed to happen is that the CO switch (or other
switch serving as the office end) will remove battery voltage from the
loop for about 250 ms within 6 seconds after the far-end party
disconnects.
As far as I can tell, most CO switches now seem to provide disconnect
supervision by default on loop-start lines. Consequently, it can be
difficult to find someone at telco who knows anything about it.
...
As for availability on residential service, just check your current
loop-start line(s) with a voltmeter and see if it drops toward zero
for about 250 ms when the far end disconnects from the call. If
there's no disconnect supervision, you'll see the voltage stay at
about 7-8 VDC when off-hook, and about 48 VDC on-hook.
[/cut]
I can hear what sounds like a battery drop on my lines when the far-end 
disconnects but asterisk doesn't seem to detect it.

Anyone have ideas.  Or experience with this?
Regards
Glen Johnson 

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Re: [Asterisk-Users] T100P Configuration for Mixed Voice & Data

> Ok, firstly HDLC is just a Layer 2 protocol like IP, so no matter what
> encapsulation they use it's still HDLC.

I meant Ethernet/ARP IP is at layer 3 DUH...
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Re: [Asterisk-Users] T100P Configuration for Mixed Voice & Data

> I need to know how to setup the data side of the T1 on my Linux Box. I
> have found information about configuring a PRI and HDLC but nothing
> about the Frame-Relay type setup for data.

Ok, firstly HDLC is just a Layer 2 protocol like IP, so no matter what
encapsulation they use it's still HDLC.

> The following is information from our T1 provider.
> Network T1:
> Framing = ESF
> Line code = B8ZS
> Build out = 0-133ft(DSX)/0dB(CSU)
> Clock = network
> Pulse-density-enforce = off
> alarm-option = on
> alarm-delay = 15
> is-slave = off
>
> DS0 Provisioning:
> analog-begin = 1
> analog-end = 16
> data-begin = 17
> data-end = 24
> alignment = same

-zaptel.conf-

span=0,0,0,esf,b8zs   ; Set up a span number 0 with the provider as
the timing source, an LBO of 0-133ft with esf framing and b8zs coding.
fxsks=1-16; basically if you're not on a PRI then
you're on a channel bank... You need to know what kind of signalling your
provider uses,
  ;  is it LoopStart, GroundStart,
E&M or KewlStart...
nethdlc=17-24; Combine channels 17-24 into data for the
Linux HDLC layer...

-zapata.conf-

signalling=fxs_ks
context=yourcontext
channel => 1-16

>From here on out, it's not an * issue, it's a Linux HDLC Layer issue which
is beyond the scope of this list... You have enough information to get it
working though, from your Vina you can see that your DLCI is 100, you'll
need the gateway address of the router on the other end and DNS information.
Also whether or not they're using PPP encapsulation, all of this is
configured with the sethdlc program and also /etc/resolv.conf and
/etc/sysconfig/network and the "ifup" scripts if you're using RedHat.

That should work... I might have the FXO/FXS thing reversed I'm always doing
that, but if it doesn't work, reverse them and it should... Like I mentioned
in my comment you need to see what kind of signalling your provider is
using, is it GroundStart, LoopStart, E&M/E&M Wink or KewlStart...

I haven't done this but I am thinking about switching to a setup much like
this so if you have success/failure let me know!

-Chris

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[Asterisk-Users] Help Me - SIP Phones ( No Voice) !!!!

2004-09-01 Thread Jefferson Carvalho

Hello list,
I've posted my problem on BSD list and i still have the
problem.
The remote side receives the call , but there's no voice
on the call.
I tried everything about possible NAT problems ..
but ther're on same net.
My platform:
FreeBSD 5.2.1-Release
Asterisk 1.0-RC2
soft phones : X-Lite

-- Executing Dial("SIP/1260-a7ae", "SIP/1262|20") in new stack
-- Called 1262
-- SIP/1262-c597 is ringing
-- SIP/1262-c597 answered SIP/1260-a7ae
-- Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
- Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
Sep  1 14:53:17 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum 
retries exceeded on call 
[EMAIL PROTECTED] for seqno 11288 
(Non-critical Response)
*
> My sip.conf

*[1260]
type=friend
username=1260
secret=jeff
context=sip
qualify=300
mailbox=1260
callerid="Jefferson Carvalho" <1260>
host=dynamic
nat=no
canreinvite=no
allow=gsm
;
[1262]
type=friend
context=sip
username=1262
secret=1262
qualify=300
callerid="Ialle" <1262>
host=dynamic
nat=no
canreinvite=no
allow=gsm
;
*>> My extensions.conf
*
[general]
static=yes
writeprotect=no
[globals]
CONSOLE => Console/dsp
IAXINFO => guest
TRUNK => Zap/g2
TRUNKMSD => 1
[sip]
exten => 1260,1,Dial(SIP/1260,20)
exten => 1261,1,Dial(SIP/1261,20)
exten => 1262,1,Dial(SIP/1262,20)
Best Regards,
-Jefferson Carvalho
IT Analist
Credishop S/A
Teresina-PI-Brasil
5586-94321901
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[Asterisk-Users] MWI light on Cisco Phones

2004-09-01 Thread Daniel Jimenez

Hi all, I'm having sudden MWI problems. Everything else on the phone 
works fine though.

I have three Cisco 7940s.
Asterisk server is behind a firewall running NAT. (192.168.1.202/24)
Phone #1 - On the same subnet 192.168.1.250. Everything works great.
Phone #2 - On a different subnet, 192.168.2.0/24. Everything works fine 
except the MWI. It never comes on. This is over an IPSEC VPN, but it's 
still behind NAT. No port or protocol restrictions.

Phone #3 - Out on the public internet (public IP), it accesses Asterisk 
through a open hole on the firewall (5034 is open, UDP). Everything 
works great except MWI never comes on.

The firewall is a PIX firewall. The only ports open to my asterisk box 
are 22(ssh) 80(www) and UDP 5034(SIP). It appears SIP works great 
because the calls function fine.

Thanks!,
--
Daniel Jimenez 
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Re: [Asterisk-Users] Rebooting Linux / Asterisk

2004-09-01 Thread Andrew Kohlsmith

On Wednesday 01 September 2004 13:41, San Singhania wrote:
> 1.How can I load asterisk automatically in Linux each time the machine
> boots up (like autoexec.bat in windows) 2.how I can shut down and restart
> asterisk automatically every night?

I use this:

su - root -c '/usr/bin/screen -d -m /usr/sbin/asterisk -g'

This puts asterisk in a screen session and allows it to receive remote 
requests (asterisk -r)

the su root is required so that the screen session shows up in root's screen 
list.

Why would you want to shut down and restart asterisk every night?

-A.
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RE: [Asterisk-Users] Rebooting Linux / Asterisk

2004-09-01 Thread Huddleston, Robert




Look 
at the /etc/rc.d and init.d directories... Unix has run levels and there are 
shell scripts (like batch files) that are called upon system 
booting..
As far 
as nightly restart - you need to use cron or atd (at daemon)  these 
processes allow you to schedule scripts to run...
Otherwise that's it - try doing some research online and if that doesn't 
help - ask offline

  -Original Message-From: San Singhania 
  [mailto:[EMAIL PROTECTED]Sent: Wednesday, September 01, 2004 1:42 
  PMTo: AsteriskSubject: [Asterisk-Users] Rebooting Linux 
  / Asterisk
  Hello everyone,
   
  I am new to Linux, some help with the following would really 
  be appreciated :
  1.How can I load asterisk automatically in Linux each time the 
  machine boots up (like autoexec.bat in windows)2.how I can shut down 
  and restart asterisk automatically every night?
  Thanks
  San
   
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[Asterisk-Users] Rebooting Linux / Asterisk

2004-09-01 Thread San Singhania




Hello everyone,
 
I am new to Linux, some help with the following would really 
be appreciated :
1.How can I load asterisk automatically in Linux each time the 
machine boots up (like autoexec.bat in windows)2.how I can shut down 
and restart asterisk automatically every night?
Thanks
San
 
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[Asterisk-Users] Dynamic dialplan

2004-09-01 Thread Juan Jose Comellas

We intend to use Asterisk with a very large dialplan (with a lot of 
functionality for 3000+ users). Each user will be able to change several of 
his parameters in the dialplan, so we will be forced to reload the diaplan 
constantly. Has anybody else any previous experience with a similar 
installation? There are some things that we'd like to know, if anybody can 
help us. These are:

- Is this something that can be done safely with Asterisk? 

- Can we have a diaplan configuration update every 5 or 10 minutes without 
service interruption? 

- What happens to new calls while the dialplan configuration is being 
reloaded? 

- What happens to active calls after the dialplan configuration is updated? 

- Can we do partial updates of the dialplan (e.g. update a specific context 
instead of the whole dialplan configuration)?

- Can Asterisk have its dialplan in a database instead of having it always in 
memory?


Thanks.

-- 
Juan Jose Comellas
([EMAIL PROTECTED])
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RE: [Asterisk-Users] Asterisk SIP between two networks

Sergio Serrano [EMAIL PROTECTED] wrote:
> SIP Provider<--->ADSL router<---localnet 
> 192.168.20.0--->ASTERISK<---localnet 172.24.240.0--->softphones
> 
> first localnet 192.168.20.0
> second localnet 172.28.240.0
> in second localnet we have softphone and the first localnet is
> connected to ADSL router to connect to our SIP provider.
> 
> if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I
> can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I
> can register in my SIP provider but softphones can't register into
> asterisk. I 'm using asterisk RC1.
> 
You probably need to use the "localnet" setting in sip.conf.  See here
for more sip.conf-related information:

http://www.voip-info.org/wiki-Asterisk+config+sip.conf

On the other hand, you could use an IAX2 provider and side-step the
issue altogether.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] NEWBIE: PWLIB Build Failure

2004-09-01 Thread Jeremy McNamara

Huddleston, Robert wrote:
Any got experience w/ PWLIB - sorry I know it's somewhat off topic... 

I do not have a bison.simple file located on Fedora RC2...
But when make'ing PWLIB I get
../common/getdate.y:106:1: warning: "YYPURE" redefined
../common/getdate.tab.c:43:1: warning: this is the location of the previous
definition

Warnings are not errors or failures.
Jeremy McNamara
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RE: [Asterisk-Users] Mitel 5010

2004-09-01 Thread slwatts


ok - just spoke to some (non-tech) at
mitel...apparently they are going to release a firmware image for the 5220
that is dual boot (Minet and SIP). One of their techs is gving me a call
back today or tomorrow (hopefully) with more details fingers crossed
it will be a downloadable item and just not hardcoded in NEW phones.

Sam

Colin Anderson <[EMAIL PROTECTED]>
wrote on 01/09/2004 16:36:24:

>  
> 
>  > I will get a packet sniffer on one in a minute  
>  
> Don't bother, won't work. I already tried. Spoke
to some Mitel mucky-mucks 
> too, and they said nope. You have to get Mitel's SIP-specific phone
which *is*
> a 5220 that's been reflashed. Unfortunately, I have 80 5020's which
can't be 
> reflashed to SIP. 
>  
> Of course, I would be *very very* interested
if you magically got it to work. 
> Different eyes, and all that... 
> 
>  
> [attachment "ATT17674.txt" deleted
by Samuel Luxford-Watts/Winckworths] 
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Parliamentary Agents 
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RE: [Asterisk-Users] Asterisk SIP between two networks

2004-09-01 Thread Sergio Serrano

Hi,
I have any information more, I have noticed that asterisk
receives 401 Unauthorized message but If I do a sip denbug I can read
next:

Sip read: 
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK6a231e4d
From: ;tag=as0f12fef4
To:
;tag=84448f3c7053227cca70775302748de3.a036
Call-ID: [EMAIL PROTECTED]: 122 REGISTER
WWW-Authenticate: Digest realm="voztele.com",
nonce="4135f73170422e2f6d1bd01c77eca25260de8f4b"
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0

If you can see next line Call-ID:
[EMAIL PROTECTED]: 122 REGISTER, Cseq
field is incompleted. How it is possible?. If I put asterisk in only one
localnet with bindaddr=192.168.20.10 I haven't the problem, but If I put
asterisk in two localnet with bindaddr=0.0.0.0 I obtain this command.

Any idea? Please, I need help

Regards,

srsergio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano
Enviado el: miércoles, 01 de septiembre de 2004 12:51
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks

Hi all, 

I have any information more. I have configured sip.conf with
bindaddr=0.0.0.0. I have observed traffic with tethereal and I have seen
the next. First REGISTER goes out from my asterisk to my SIP Provider.
My SIP Provider respond to my with a 401 Unauthorized meesage, but
Asterisk doesn't read this message and try to resend first REGISTER.

In the second localnet(in previous message) there is a Hicom Siemens
with a HG1500 interface with Intel propietary protocol, but without SIP
protocol. I have noticed that this interface goes down when I start
asterisk.

Has anyone had same problem? Could anyone help me with this
problem?

Best regards,
srsergio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano Enviado el: miércoles, 01 de septiembre de 2004 0:46
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: [Asterisk-Users] Asterisk SIP between two networks

Hi all,
I have next configuration:

SIP Provider<--->ADSL router<---localnet
192.168.20.0--->ASTERISK<---localnet 172.24.240.0--->softphones

first localnet 192.168.20.0
second localnet 172.28.240.0
in second localnet we have softphone and the first localnet is
connected to ADSL router to connect to our SIP provider.

The problem is the next:
if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I
can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I
can register in my SIP provider but softphones can't register into
asterisk. I 'm using asterisk RC1. 

Any idea?

srsergio

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Re: [Asterisk-Users] Can asterisk detect BUSY signal?

2004-09-01 Thread Brian Capouch

Andrew Kohlsmith wrote:
On Tuesday 31 August 2004 17:36, Kevin Walsh wrote:
Spam-dialling should be made illegal.  I, for one, wouldn't spend two
seconds adding features to support this sort of usage.

I can think of at least one legitimate use for this -- reverse spam dialling, 
or at least "real person" detection.  I hate sitting in hold queues and my 
usual method is to put the phone on speaker and listen to Muzak while I wait.

Wouldn't it also be a legtimate use when instances arise (I had one two 
weeks ago) where one would need to contact a set (or subset) of 
customers in a timely fashion?

We had to do an emergency router swap, and something like that would 
sure have been nice compared to the way I had to do it.

Not all automated dialing is evil.
B.
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Re: [Asterisk-Users] DeadAGI Application

2004-09-01 Thread Darren Wiebe

I'm not sure what the problem is with my source.  I'm running the latest 
copy, as of this morning, from cvs.  DeadAGI will not build on my server 
but it will build on my laptop.  I'll have to have a look at my source.  
I just copied the res_agi.so file across this morning and everything 
works fine.  Thanks for your help.

Darren Wiebe
[EMAIL PROTECTED]
Florian Overkamp wrote:
Hi, 

 

-Original Message-
I downloaded the astcc calling card program.  Thanks, it is 
very easy to 
setup and works Excellent.  Anyway, it says to use DeadAGI to run it 
rather than AGI.  I don't know what I am doing wrong.  I just 
updated my 
asterisk from cvs and rebuilt and reinstalled.  I do not have an 
application called DeadAGI.  I have searched the source, google, etc. 
but have not been able to find anything.  Any pointers?
   

It should be in the regular AGI code, but tell us, what version of asterisk
are you running ? DeadAGI has been around for a while, but not forever, so
if your code is 'ancient', better upgrade :-)
Best regards,
Florian
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[Asterisk-Users] NEWBIE: PWLIB Build Failure

2004-09-01 Thread Huddleston, Robert

Any got experience w/ PWLIB - sorry I know it's somewhat off topic... 

I do not have a bison.simple file located on Fedora RC2...

But when make'ing PWLIB I get

../common/getdate.y:106:1: warning: "YYPURE" redefined
../common/getdate.tab.c:43:1: warning: this is the location of the previous
definition

Is it safe to continue or is this a bad build?
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RE: [Asterisk-Users] All you polycom folks.....

2004-09-01 Thread Chris HARIGA

http://www.freedomphones.net/polycom/files/

Best regards,

Chris HARIGA

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven Kokinos
Sent: Wednesday, September 01, 2004 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] All you polycom folks.

Does anyone know where to obtain the 2.5 / 1.3.1 bootrom/app? My 
account on polycom's site keeps pointing me at documentation only.

Regards,

-Steve

On Sep 1, 2004, at 10:00 AM, Matthew Marlowe wrote:

> I'm using the same SIP version, everything is running great except as 
> I've said before that setting the default ring type isn't working and 
> incoming calls only displays name and not name and number..
>
> 
>
> From: [EMAIL PROTECTED] on behalf of Reid A. 
> Forrest
> Sent: Wed 9/1/2004 10:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] All you polycom folks.
>
>
>
> I'm using SIP 1.3.1 with Boot RPM 2.50 and so far everything's running 
> great.
> I don't use # transfer though, so haven't tried that. I use the 
> softkeys
> instead to transfer.
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of
>> Matthew Marlowe
>> Sent: Wednesday, September 01, 2004 8:57 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: RE: [Asterisk-Users] All you polycom folks.
>>
>> I left my phone at home I think Im using sip 1.3.1.. It's 1.3.
>> something
>>
>> Asterisk CVS-HEAD-05/12/04-13:23:20, Copyright (C) 1999-2004 Digium.
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Brent
>> Franks
>> Sent: Tuesday, August 31, 2004 10:14 PM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] All you polycom folks.
>>
>> Just out of curiosity,
>>
>> What version of CVS and Polycom SIP software are you running happily?
>>
>> Are you running SIP 2.3.0 yet?  2.2.0?  2.1.0?
>>
>> I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1
>> with poor results.  Transferring did not work as expected.
>> Using the #
>> key to do blind transfers after a call was on hold did not work.
>>
>> Just curious.
>>
>> Thanks,
>>
>> - Brent
>>
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[Asterisk-Users] Newbie - Troubles after installing e100p

2004-09-01 Thread Matthias Leeb

Hello

I's trying to install an asterisk-server with a wildcard e100p and get
some errors while starting asterisk:

Sep  1 18:49:34 WARNING[1024]: chan_zap.c:724 zt_open: Unable to specify
channel 1: No such device or address
Sep  1 18:49:34 ERROR[1024]: chan_zap.c:5883 mkintf: Unable to open
channel 1: No such device or address here = 0, tmp->channel = 1, channel
= 1
Sep  1 18:49:34 ERROR[1024]: chan_zap.c:8792 setup_zap: Unable to
register channel '1'

The wildcard is connected to an local pstn. I first had troubles with
the signalling after solving this, the failures came up instead...

I'm quit sure that i'm doing an obvious failure but even after reading
the wiki-pages for hours i'm not able to find the error.

Best regards

matthias

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Re: [Asterisk-Users] Asterisk codecs and packet size

2004-09-01 Thread Andres


The quick and dirty way:

In rtp.c, function "ast_rtp_write", in the "switch" statement,
"AST_FORMAT_G729A" case, change the smoother creation to something
larger. E.g.:
rtp->smoother = ast_smoother_new(40);
Keep in mind that you must set this into something valid
(45 obviously is not valid). Recompile and you should be fine.
Michael, this little nugget made my day.  Last year we offered to pay 
for this development.  Too bad you didn't collect:)

Thanks!
--
Andres
Network Admin
http://www.telesip.net
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Re: [Asterisk-Users] SMS & Asterisk - an explanation

2004-09-01 Thread Asterisk

Oh great. BT at it's best.

Spoken to 4 different product managers / isdn help desk / customer service
weenies. Each one says "Eh? What?" and I've got to point them to your link.
"I'll get back to you on that" is the next response.

Still waiting 

But thanks for the link - without that I would have not stood a change
against corporate inertia / stupidity.

Julian

- Original Message - 
From: "Scott Stingel" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Wednesday, September 01, 2004 3:23 PM
Subject: RE: [Asterisk-Users] SMS & Asterisk - an explanation


> Hi Julian-
>
> I was using a BT BRI line, with caller ID option enabled.  Also, I had to
> send 1470 before the call because my customer had blocked his outgoing
> number on this line.
>
> So I'm certain that it works on BRI.  BT says in their SIN document
> (Supplier's Information Note), number 413, that analogue lines and both
ISDN
> 2e and 30e can provide this service.  See paragraph 3.2 of this document.
>
> Here's the link:  http://www.sinet.bt.com
>
>
> Regards
> Scott
>
>
> Scott M. Stingel
> President,
> Emerging Voice Technology, Inc.
> Palo Alto California & London England
> www.evtmedia.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
> Sent: Wednesday, September 01, 2004 12:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SMS & Asterisk - an explanation
>
> I tried to send sms messages the other day from a * box connected to a E1
> line (BT ISDN30).
>
> Message never arrived, however, I was soon called back on the E1 by an
> automated BT system which sent a message stating that "you cannot send sms
> messages on this line"
>
> Is there anything I need to do before I start sending text messages ? Is
it
> the ISDN30 that is the problem, and do I need to send SMS via standard
lines
> (pots) or ISDN2e lines ?
>
> Julian.
>
>
>
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RE: [Asterisk-Users] Mitel 5010

2004-09-01 Thread Colin Anderson




 

   > I will get a packet sniffer on one 
  in a minute  
   
  Don't bother, won't work. I already tried. Spoke to some Mitel 
  mucky-mucks too, and they said nope. You have to get Mitel's SIP-specific 
  phone which *is* a 5220 that's been reflashed. Unfortunately, I have 80 
  5020's which can't be reflashed to SIP. 
   
  Of 
  course, I would be *very very* interested if you magically got it to work. 
  Different eyes, and all that... 
   
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[Asterisk-Users] CLI variable not set on incoming call

2004-09-01 Thread Alessio Focardi

Hi,

need a quick help ... it should be easy but ...

exten =>_9898,1,Answer
exten =>_9898,2,VoiceMailMain([EMAIL PROTECTED])


 Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer("Zap/8-1", "") in new stack
-- Executing VoiceMailMain("Zap/8-1", "@domain") in new stack



As you can see there variable CALLERID is empty, why ?

I tried also with CALLERIDNUM, same result.

Tnx for any help .




-- 
Best regards,
 Alessio  mailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] All you polycom folks.....

2004-09-01 Thread Steven Kokinos

Does anyone know where to obtain the 2.5 / 1.3.1 bootrom/app? My 
account on polycom's site keeps pointing me at documentation only.

Regards,
-Steve
On Sep 1, 2004, at 10:00 AM, Matthew Marlowe wrote:
I'm using the same SIP version, everything is running great except as 
I've said before that setting the default ring type isn't working and 
incoming calls only displays name and not name and number..


From: [EMAIL PROTECTED] on behalf of Reid A. 
Forrest
Sent: Wed 9/1/2004 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] All you polycom folks.


I'm using SIP 1.3.1 with Boot RPM 2.50 and so far everything's running 
great.
I don't use # transfer though, so haven't tried that. I use the 
softkeys
instead to transfer.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Marlowe
Sent: Wednesday, September 01, 2004 8:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] All you polycom folks.
I left my phone at home I think Im using sip 1.3.1.. It's 1.3.
something
Asterisk CVS-HEAD-05/12/04-13:23:20, Copyright (C) 1999-2004 Digium.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent
Franks
Sent: Tuesday, August 31, 2004 10:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] All you polycom folks.
Just out of curiosity,
What version of CVS and Polycom SIP software are you running happily?
Are you running SIP 2.3.0 yet?  2.2.0?  2.1.0?
I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1
with poor results.  Transferring did not work as expected.
Using the #
key to do blind transfers after a call was on hold did not work.
Just curious.
Thanks,
- Brent
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RE: [Asterisk-Users] Mitel 5010

2004-09-01 Thread slwatts


Fair enough, The MItel is not the most
expensive VOIP solution, but as you say - the reason Asterisk is so attractive
is its effectively free and you are not locked to a single vendor or support
channel.

I have only just started looking at
Asterisk, I like it a lot, particularly its openness - the Mitel solution
is an almost watertight black box :-(

I would still like to use the 5220's
with asterisk (they cost about the same for us as the cisco 7960's) but
are smaller, neater and look more like a phone (surprisingly!)

I will get a packet sniffer on one in
a minute

Sam

Colin Anderson <[EMAIL PROTECTED]>
wrote on 01/09/2004 15:24:50:

> Ditto that. On a Mitel box, A Service Contract Is Your Friend and
a Good 
> Interconnect Is Your Best Friend. Our 3300 was loosing conn w/ the
phones and 
> loosing sync with our PRI. Our interconnect blamed everything from
the LAN to 
> sunspots except the 3300. Because we had a service contract, I insisted
that 
> the NSU be replaced, which they did grudgingly and lo! problems ceased.
~3k 
> calls / day from 80 users for 3 years so far - no problems. 
>  
> FWIW, I'm just using QOS. We haven't had any
issues with non - VLAN config. 
> However, the reason I'm on this list in the first place is because
we are 
> planning to replace the 3300 with * :-) and the reason is simply because
the 
> Mitel box is too expensive $$. A 5020 costs us $1200 Cdn. My
boss, who 
> approved the system in the first place, likes to tease me and say:
"Tell me 
> again why we moved away from a Meridian One?"
>  
> On topic: The Ipera (the 3300's predecessor)
supported SIP, as well as the 
> new, I think it's called 3010 which is throttled to 25 users. Mitel
made a 
> stupid decision to drop SIP for the 3300 to try to lock in users with
their 
> own proporietary protocol which I have analyzed and it's part IP and
part CDS 
> (Cisco Discovery, like SNMP). Because the CDS part is Layer - 2 -
ish it's 
> hard to route; my BSD box routes it OK but a Seawall kind of config
won't. G
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, September 01, 2004 7:20 AM
> To: Asterisk Users Mailing List - Non-Commercial
Discussion
> Subject: Re: [Asterisk-Users] Mitel 5010
> 
> I am very surprised that you have had problems with a mitel 3300ICP.
We have 
> had one installed here for the past year or so and it is a very good
system. 
> We did suffer the occaisonal crash and then our system was wiped out
by a 
> lightning strike, we had a replacement controller in place and fully
working 
> within 3 hours. Since then we have had absolutely no crashes at all.

> 
> You do need to setup the 3300ICP properly - incorrect configuration
can cause 
> you serious nightmares, but then it is a very flexible system. It
also seems 
> to be very particular about your network setup. Dont even think about
trying 
> to run it over a data network that does not have vlans or QOS. 
> 
> The early software versions were pretty poor, however version 4 is
much better
> and we will soon be upgrading to version 5 so if you want to know
how that 
> fares let me know. 
> 
> Their teleworker solution is particularly impressive, although I wish
they 
> would release SIP support for their phones and controllers. 
> 
> Anyone know of a project to bring minet to asterisk? 
> 
> Sam 
> 
> Stephen Stull <[EMAIL PROTECTED]> wrote on 01/09/2004 13:33:59:
> 
> > Is this the 3300IP system?  Those systems *can* be quite
good, but 
> > need to be up to the latest revision and
require a bit of tuning.  I'd 
> > be interested in your finds regardless and
could probably experiment 
> > with a 5005, 5010 or even the higher-end
ones. 
> > 
> > I'll ask a former co-worker if I can borrow
one for testing. 
> > ___

> > Asterisk-Users mailing list 
> > [EMAIL PROTECTED] 
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> >    http://lists.digium.com/mailman/listinfo/asterisk-users

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b

RE: [Asterisk-Users] Group Dial

Tomica Crnek [EMAIL PROTECTED] wrote:
> (Article auto-converted from unnecessary HTML to nice plain text.)
> 
> I want to have a group and dial multiple phones/lines simultaneously. If
> I use this Dial command: 
> 
> exten => 222,2,Dial(${TRUNKBP}/246&SIP/258&${TRUNKBP}/243,20,tTr)
> 
> ... all phones ring just once, after that only the first one continues
> ringing and only that one can answer. Can anyone tell me why? 
> 
I haven't noticed that on my setups; all phones ring as expected.  Are
you using the latest CVS version or some old version.  Perhaps an
upgrade will help.

By the way, I don't use the [tTr] flags either, but I don't think that
makes a difference in this case.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] Using regular expression in dialplan

2004-09-01 Thread Selim

Sorry for the re-posting, I received a delivery failure mail for each
one of my emails ...
Thank you Kevin for your "Clues".

Does any one else have used regex (or extended regex) with GotoIf ?

Selim

On Wed, 1 Sep 2004 14:34:49 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Selim [EMAIL PROTECTED] wrote:
> > Did anyone manage to make the GotoIf command work with regular expression
> >
> Does re-posting the same question lots of times usually work for you?
> I suppose it generated a response in this case.  I usually just delete
> duplicates, along with the original.
> 
> >
> > I've tried the follwoing syntax but it is alway going to 2 whatever
> > the value of DTMFSeq:
> >
> > exten => s,1,GotoIf($[${DTMFSeq} : 123]?s|4:s|2)
> > 
> Clue: 123 is 123.  It's not a regular expression that would match 1, 2
> or 3.  123 will match 123, and nothing else.  [Perhaps something is
> missing from your expression.]
> 
> >
> > The only way I managed to make it work is the following :
> >
> > exten => s,1,GotoIf($[${DTMFSeq} : 1|2|3]?s|4:s|2)
> >
> Clue: 1|2|3 is a regular expression that would match 1, 2 or 3.  That's
> why it works.
> 
> --
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Re: [Asterisk-Users] Lucent iMerge

2004-09-01 Thread Josh Krueger

It may work if you can make the iMerge register with Asterisk, currently
( to the extent of my knowledge ) mgcp can only act as the registrar.

On Wed, 2004-09-01 at 06:59, Huddleston, Robert wrote:
> I've read the wiki and other resources on how to connect Vonage / Voicepulse
> and all these other services to Asterisk... We are attempting a connection
> to a Lucent iMerge. Lucent has told us that it won't work - but we feel
> confident that it will. Has anyone worked with the Lucent iMerge - or would
> be willing to help lend a hand?
> It is capable of H323 / MGCP. Even if I could make the Asterisk register
> with the iMerge I would be happy enough to now that I might be able to make
> this system work.
> 
> Please help... Thanks
> 
> Robert A. Huddleston, KF4BYY
> Cavalier Telephone LLC.
> 804.422.4401
> [EMAIL PROTECTED]  
>  
> 
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[Asterisk-Users] Agents Log off

2004-09-01 Thread João Amaro

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
I'm using the apllication AgentCallBackLogin so agents can login to a
queue. They just need to
enter the password, the CallBack Extensions is the ${CALLERIDNUM}
Is there a way to AgentsLogOff withou using the AgentCallBackLogin
application. I don't want the
user to enter they CALLERIDNUM.
Regards

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFBNd4xJUm/Bor63CERAm6xAJ9EZcU6B1CRDfyHQVKmsnEFqegFBgCeJWpG
05VztJM3tj3mEiuOG4Lk+DU=
=SjNj
-END PGP SIGNATURE-
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RE: [Asterisk-Users] SMS & Asterisk - an explanation

2004-09-01 Thread Scott Stingel

Hi Julian-

I was using a BT BRI line, with caller ID option enabled.  Also, I had to
send 1470 before the call because my customer had blocked his outgoing
number on this line.

So I'm certain that it works on BRI.  BT says in their SIN document
(Supplier's Information Note), number 413, that analogue lines and both ISDN
2e and 30e can provide this service.  See paragraph 3.2 of this document.

Here's the link:  http://www.sinet.bt.com


Regards
Scott 


Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Wednesday, September 01, 2004 12:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SMS & Asterisk - an explanation

I tried to send sms messages the other day from a * box connected to a E1
line (BT ISDN30).

Message never arrived, however, I was soon called back on the E1 by an
automated BT system which sent a message stating that "you cannot send sms
messages on this line"

Is there anything I need to do before I start sending text messages ? Is it
the ISDN30 that is the problem, and do I need to send SMS via standard lines
(pots) or ISDN2e lines ?

Julian.



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RE: [Asterisk-Users] Mitel 5010

2004-09-01 Thread Colin Anderson

Ditto
that. On a Mitel box, A Service Contract Is Your Friend and a Good Interconnect
Is Your Best Friend. Our 3300 was loosing conn w/ the phones and loosing sync
with our PRI. Our interconnect blamed everything from the LAN to sunspots except
the 3300. Because we had a service contract, I insisted that the NSU be
replaced, which they did grudgingly and lo! problems ceased. ~3k calls / day
from 80 users for 3 years so far - no problems.
FWIW,
I'm just using QOS. We haven't had any issues with non - VLAN config. However,
the reason I'm on this list in the first place is because we are planning to
replace the 3300 with * :-) and the reason is simply because the Mitel box is
too expensive $$. A 5020 costs us $1200 Cdn. My boss, who approved the
system in the first place, likes to tease me and say: "Tell me again why we
moved away from a Meridian One?"
On
topic: The Ipera (the 3300's predecessor) supported SIP, as well as the new, I
think it's called 3010 which is throttled to 25 users. Mitel made a stupid
decision to drop SIP for the 3300 to try to lock in users with their own
proporietary protocol which I have analyzed and it's part IP and part CDS (Cisco
Discovery, like SNMP). Because the CDS part is Layer - 2 - ish it's hard to
route; my BSD box routes it OK but a Seawall kind of config won't.
G

-Original Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]Sent: Wednesday, September 01,
2004 7:20 AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Mitel
5010I am very surprised
that you have had problems with a mitel 3300ICP. We have had one installed
here for the past year or so and it is a very good system. We did suffer the
occaisonal crash and then our system was wiped out by a lightning strike, we
had a replacement controller in place and fully working within 3 hours. Since
then we have had absolutely no crashes at all. You do need to setup the 3300ICP properly - incorrect
configuration can cause you serious nightmares, but then it is a very flexible
system. It also seems to be very particular about your network setup. Dont
even think about trying to run it over a data network that does not have vlans
or QOS. The early software
versions were pretty poor, however version 4 is much better and we will soon
be upgrading to version 5 so if you want to know how that fares let me
know. Their teleworker solution is
particularly impressive, although I wish they would release SIP support for
their phones and controllers. Anyone know of a project to bring minet to asterisk?
Sam Stephen Stull <[EMAIL PROTECTED]> wrote on 01/09/2004
13:33:59:> Is this the 3300IP system? Those systems *can* be
quite good, but > need to be up to the
latest revision and require a bit of tuning. I'd > be interested in your finds regardless and could probably
experiment > with a 5005, 5010 or even the
higher-end ones. > > I'll ask a
former co-worker if I can borrow one for testing. > ___
> Asterisk-Users mailing list > [EMAIL PROTECTED] >
http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >
http://lists.digium.com/mailman/listinfo/asterisk-users
Winckworth Sherwood Solicitors and
Parliamentary Agents DX 148400 WESTMINSTER 5 : 35 Great Peter Street,
London SW1P 3LR Telephone 020 7593 5000 Fax 020 7593 5099
Confidentiality This email message and any attachments are
confidential; they may be subject to legal professional privilege and are
intended for the named recipient only. If you are not the named recipient,
please return the message and enclosures immediately and delete them from your
system.
Caution Before advice received only by email (whether by attachment
or otherwise) may be relied on, the authenticity of the communication must be
verified by means independent of email.
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Partners A list of partners is available for inspection at each
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[Asterisk-Users] Group Dial

2004-09-01 Thread Tomica Crnek




Hi 
everyone,
 
I want to have a 
group and dial multiple phones/lines simultaneously. If I use this Dial 
command:
 
exten => 
222,2,Dial(${TRUNKBP}/246&SIP/258&${TRUNKBP}/243,20,tTr)
 
... all phones 
ring just once, after that only the first one continues ringing and only that 
one can answer. Can anyone tell me why?
 
thanks!
Tomica
 
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RE: [Asterisk-Users] All you polycom folks.....

2004-09-01 Thread Matthew Marlowe

I'm using the same SIP version, everything is running great except as I've said before 
that setting the default ring type isn't working and incoming calls only displays name 
and not name and number..



From: [EMAIL PROTECTED] on behalf of Reid A. Forrest
Sent: Wed 9/1/2004 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] All you polycom folks.



I'm using SIP 1.3.1 with Boot RPM 2.50 and so far everything's running great.
I don't use # transfer though, so haven't tried that. I use the softkeys
instead to transfer.

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Matthew Marlowe
> Sent: Wednesday, September 01, 2004 8:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] All you polycom folks.
>
> I left my phone at home I think Im using sip 1.3.1.. It's 1.3.
> something
>
> Asterisk CVS-HEAD-05/12/04-13:23:20, Copyright (C) 1999-2004 Digium.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Brent
> Franks
> Sent: Tuesday, August 31, 2004 10:14 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] All you polycom folks.
>
> Just out of curiosity,
>
> What version of CVS and Polycom SIP software are you running happily?
>
> Are you running SIP 2.3.0 yet?  2.2.0?  2.1.0?
>
> I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1
> with poor results.  Transferring did not work as expected. 
> Using the #
> key to do blind transfers after a call was on hold did not work.
>
> Just curious.
>
> Thanks,
>
> - Brent
>
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RE: [Asterisk-Users] All you polycom folks.....

2004-09-01 Thread Reid A. Forrest

I'm using SIP 1.3.1 with Boot RPM 2.50 and so far everything's running great.
I don't use # transfer though, so haven't tried that. I use the softkeys
instead to transfer. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matthew Marlowe
> Sent: Wednesday, September 01, 2004 8:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] All you polycom folks.
> 
> I left my phone at home I think Im using sip 1.3.1.. It's 1.3.
> something
> 
> Asterisk CVS-HEAD-05/12/04-13:23:20, Copyright (C) 1999-2004 Digium. 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Brent
> Franks
> Sent: Tuesday, August 31, 2004 10:14 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] All you polycom folks.
> 
> Just out of curiosity,
> 
> What version of CVS and Polycom SIP software are you running happily?
> 
> Are you running SIP 2.3.0 yet?  2.2.0?  2.1.0?
> 
> I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1
> with poor results.  Transferring did not work as expected.  
> Using the #
> key to do blind transfers after a call was on hold did not work.
> 
> Just curious.
> 
> Thanks,
> 
> - Brent
> 
> ___
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RE: [Asterisk-Users] Using regular expression in dialplan