date:20050130

[Asterisk-Users] Re: Record inbound and outbound calls to and from one number.

2005-01-30 Thread Tom Shoval

Tim Mattison wrote:

> Good call.
> 
> For our American readers... does anyone know where I can obtain a list
> of states/counties and their regulations in regards to call recording?
> 
> On Sun, 2005-01-30 at 22:10 +, Mike Dent wrote:
> > or maybe country? or should that be County? :)
> > 
> > Mike
> > 
does it matter?
you should provide warning to everybody.

you can do it in your top menu, by stating that "some calls are
monitored for QA purposes", like most call centers do these days anyway.


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Re: [Asterisk-Users] Japan

2005-01-30 Thread Jason Frisch

Eicon Diva Server BRI = ISDN I think..
JAson
Leo Ann Boon wrote:

Jason Frisch wrote:
Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
finding hardware that is supported. I tried
Voicetronix but they said that they are too busy
to create a driver.

I knew someone in Japan who had a working Asterisk + Eicon Diva Server 
BRI setup.


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Re: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe

2005-01-30 Thread Remco Barende

On Sun, 30 Jan 2005, Martin List-Petersen wrote:
Citat Remco Barende <[EMAIL PROTECTED]>:
I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything
seems to be running fine but after some time asterisk just goes crazy
(even withouth any incoming or outgoing call activity perviously).
If I leave the box up for some time * goes haywire and the console is
flooded with this message:
Ouch ... error while writing audio data: : Broken pipe
At that time I can see that there are multiple instances of mpg123 active.
The solution to this problem is to kill-9 mpg123, do the same for *,
unload the modules and then load the modules again and start asterisk. If
I do not unload re-load the modules I cannot access the ISDN line nor do
incoming calls work.
I really don't know where to look for this problem. Is it possible to
completely disable music on hold? Asterisk combined mpg123 is causing
nothing but problems anyway, the current stable still leaves abandoned
mpg123 processes.
It doesn't work :( Asterisk doesn't go haywire flooding the console but 
now simply bombs out with :

*CLI>
Segmentation fault
I guess that qualifies it as a bristuff bug?
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Re: [Asterisk-Users] Japan

2005-01-30 Thread Leo Ann Boon


Jason Frisch wrote:
I asked Softbank and it seems that using SIP etc directly is not an 
option.
Something to do with theVoIP-TA being used for communications between
the providers "call-agent".
Sounds like they're using MGCP. At this point, Asterisk is not able to 
act as an MGCP endpoint, it can only be a 'call agent'.

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Re: [Asterisk-Users] Japan

2005-01-30 Thread Leo Ann Boon


Jason Frisch wrote:
Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
finding hardware that is supported. I tried
Voicetronix but they said that they are too busy
to create a driver.
I knew someone in Japan who had a working Asterisk + Eicon Diva Server 
BRI setup.


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[Asterisk-Users] Trunked IAX or not

2005-01-30 Thread Spencer Nassar

Has anyone benchmarked Asterisk on a dedicated single versus dual
processor machine?
http://www.astertest.com/
Cheers, Philipp
The test results that Philipp pointed out show some protocol 
comparisons that include "iax2 trunking / alaw" and "iax2 / alaw" and 
concludes that "IAX2 trunking is more than twice as fast as non 
trunking IAX."

Forgive the newbie question, but what is this distinction?  In what 
cases is a connection 'trunking' or 'not'?  If I have a "register =>" 
statement in my iax.conf file, is that a trunked connection to my DiD 
provider?

Thanks!
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Re: [Asterisk-Users] PRI for Data and Voice

2005-01-30 Thread clive

Dave, howzit

You can use asterisk with a quad E1 card to divide your E1. So
anyone who dials in using 1234 for example, route to your
portmaster and anyone who dials in using 1235 use for IVR/voip,
whatever.

Good luck
Regards
Clive

On 29 Jan 2005 at 15:11, David Norton wrote:

>
> Hi,
>
> Currently I only have 1 PRI which I am using for dial-in customers. The line 
> is connected to a
> Portmaster3. I have never used more than 10 concurrent channels. The calls 
> can be both analog
> or ISDN. It would be a waste to order another PRI for my Asterisk box. Is 
> there any way of splitting a PRI into 2 PRIs of 15 channels each, or 
> plugging the PRI into the *
> box and it send the data calls to the portmaster, or handles them itself?
>
> Any advice would be much appreciated
>
> Regards
>
> David Norton
>


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Re: [Asterisk-Users] Japan

2005-01-30 Thread Jason Frisch

I asked Softbank and it seems that using SIP etc directly is not an option.
Something to do with theVoIP-TA being used for communications between
the providers "call-agent".
Jason
Steven Critchfield wrote:
On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote:
 

Sorry for my ignorance, but what is J1? I actually hope to use Softbanks 
fiber-based IPtel
service, but I believe they require VoIP TA so I guess the end result is 
just a standard
analog line.
   

J1 is a Japanese T1 or close equivalent.
If IPTel uses a VoIP TA(voice over IP terminal adapter), you might want
to check into the type of signaling they use. It sounds like they might
be using SIP, H323, or MGCP to deliver the service. In that case you
might be able to swap the TA for asterisk with no or minimal trouble.
Then you are free to provision your side of the link as you wish.
 

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Re: [Asterisk-Users] DIAX softphone - Asterisk server rejecting

2005-01-30 Thread Dan

Hi,
- Original Message - 
From: "Pradhip KCL" <[EMAIL PROTECTED]>
Hi all
trying some basic functinality
I was able  to download asterisk and complie.
I have downloaded DIAX soft phone also. I am trying to make an
internal call between two softphones.
I read the http://www.voip-info.org/wiki-Asterisk+installation+tips
website to configure channels and extensions, Since DIAX used IAX.conf
and extension.conf I tried to confifg as much as i understood. It
seems like any call  i make asterisk is not acceptiong.
It is Rejecting saying socker read error.  any help will be appreciated.
I am not sure what i configured is right also? how to proceed
Thanks & Regards
Pradhip
Try the examples from the DIAX help file.
BR,
Dan
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Re: [Asterisk-Users] how to stop ringing after congestion.

2005-01-30 Thread el Flynn

Jon Gabrielson wrote:
When there are no zap channels available, I signal congestion
using the following:
exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXX,2,Playtones(congestion)
exten => _9NXX,3,Congestion
The congestion sound plays correctly, but the ringing continues
in the background.  Why is it still ringing and how do I make it stop?
try
exten => _9NXX,3,Congestion(5)
which will stop the tones after 5 seconds.
flynn
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Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

2005-01-30 Thread Paradise Dove

On Sun, 30 Jan 2005 18:14:38 -0600, Jon Gabrielson
<[EMAIL PROTECTED]> wrote:
> Asterisk should be able to do this, there are several cases
> when this is essential.  The first is a shared/party line where
> asterisk cannot have guaranteed access for whatever reason.
> In our case, that reason happens to be because we also use
> our outgoing lines for faxing.
> The second is that without dialtone detection, if for some
> reason the line is down, asterisk needs to know so that it can
> try a different outgoing line.  If the first line is down, asterisk
> shouldn't hang, it should wait a few seconds and try to dial
> out on the next line.

this is the feature which other PBXs have.  the ability to detect out
of order lines (no dialtone - used by others) .
btw, as i said before a feature request about this is submitted to bug
tracker at http://bugs.digium.com/bug_view_page.php?bug_id=0002612.
suppose that any followups could be done there. maybe setting bounty
on this issue speedup the process!

thanks,
Paradise Dove

> 
> 
> Jon.
> 
> 
> On Sunday 30 January 2005 04:13 pm, Steven Critchfield wrote:
> > On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote:
> > > Can't asterisk look for a dialtone?  Even a $5 modem
> > > can detect whether or not there is a dialtone.
> >
> > Maybe you should just use your $5 modem and write your own software.
> >
> > Asterisk is a PBX. PBXs shouldn't have to deal with your bastardized
> > setup that doesn't respect the normal way in which a PBX is set up. A
> > PBX sits between the PSTN and ALL other access to the PSTN. In doing so,
> > asterisk can know ahead of time that the line is available. If you wait
> > for dialtone detection, then you have to also make code to understand
> > all international dialtones as well. Then you have to delay dial till
> > you are certain it is the tone you are expecting.
> >
> > > On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote:
> > > > > When I place a call with asterisk, asterisk will try to dial
> > > > > out on the first line even if the first line is already being
> > > > > used by someone else.  Any ideas on what I'm doing
> > > > > wrong?
> > > >
> > > > My question would be, how would asterisk know the line is in use if it
> > > > isn't controlling it?
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[Asterisk-Users] how to stop ringing after congestion.

2005-01-30 Thread Jon Gabrielson

When there are no zap channels available, I signal congestion
using the following:

exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXX,2,Playtones(congestion)
exten => _9NXX,3,Congestion


The congestion sound plays correctly, but the ringing continues
in the background.  Why is it still ringing and how do I make it stop?


Thanks,


Jon.
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Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips

2005-01-30 Thread jurgen

Hi Howard,

Which provider are you with? We're with Primus Business here in
Melbourne, and haven't had anything like what you're describing. For
reference, here's a snip of my zapata.conf:

[channels]

language=en
context=local
signalling=fxs_ks
usecallerid=no
echocancel=yes
echocancelwhenbridged=yes
busydetect=yes
busycount=5

Sometimes the busydetect hack hits a false positive and disconnects
during a conversation, so I'm thinking of upping the busycount, but
aside from that, calls through this are quite reliable.

Best,

...jurgen


On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote:
> Is anyone having/had a problem with a TDM400P card hanging up on STD
> outbound calls as soon as the called party answers.
> 
> I'm guessing that * is responding to the STD pips in some way.
> 
> --
> Howard.
> LANNet Computing Associates;
> Your Linux people 
> --
> "When you just want a system that works, you choose Linux;
> when you want a system that just works, you choose Microsoft."
> --
> "Flatter government, not fatter government;
> Get rid of the Australian states."
> 
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[Asterisk-Users] Zap channels in AU hanging up on STD pips

2005-01-30 Thread Howard Lowndes

Is anyone having/had a problem with a TDM400P card hanging up on STD
outbound calls as soon as the called party answers.

I'm guessing that * is responding to the STD pips in some way.

-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] Japan

2005-01-30 Thread Steven Critchfield

On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote:
> Sorry for my ignorance, but what is J1? I actually hope to use Softbanks 
> fiber-based IPtel
> service, but I believe they require VoIP TA so I guess the end result is 
> just a standard
> analog line.

J1 is a Japanese T1 or close equivalent.

If IPTel uses a VoIP TA(voice over IP terminal adapter), you might want
to check into the type of signaling they use. It sounds like they might
be using SIP, H323, or MGCP to deliver the service. In that case you
might be able to swap the TA for asterisk with no or minimal trouble.
Then you are free to provision your side of the link as you wish.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Monitor calls timeout

2005-01-30 Thread jurgen

On Mon, 31 Jan 2005 09:26:05 +0800, el Flynn <[EMAIL PROTECTED]> wrote:

> did you try setting using AbsoluteTimeout in the context? e.g.
> 
> exten => s,1,Answer
> exten => s,2,AbsoluteTimeout(0)
> exten => s,3,Monitor(wav,testrecod,m)

Thanks for the suggestion, but it's no good. It still times out after
10 seconds. It seems to be something in the Monitor application,
rather than anywhere else. I can playback a sound (like the monkeys,
or MOH) forever and ever without timing out. Monitoring kills itself
though.

Oh - and using Monitor the way it's "supposed" to be used works just
fine, with no problems or timeouts (I have one of the Zap channels set
to record everything, and that works all the time).

jurgen

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[Asterisk-Users] DIAX softphone - Asterisk server rejecting

2005-01-30 Thread Pradhip KCL

Hi all
trying some basic functinality

I was able  to download asterisk and complie.
I have downloaded DIAX soft phone also. I am trying to make an
internal call between two softphones.

I read the http://www.voip-info.org/wiki-Asterisk+installation+tips
website to configure channels and extensions, Since DIAX used IAX.conf
and extension.conf I tried to confifg as much as i understood. It
seems like any call  i make asterisk is not acceptiong.
It is Rejecting saying socker read error.  any help will be appreciated.
I am not sure what i configured is right also? how to proceed
Thanks & Regards
Pradhip
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Re: [Asterisk-Users] detailed asterisk howto

2005-01-30 Thread Pradhip KCL

http://www.voip-info.org/wiki-Asterisk+installation+tips
this might help you


On Mon, 31 Jan 2005 14:49:25 +1100, Duane <[EMAIL PROTECTED]> wrote:
> szj wrote:
> > Hi, all:
> >
> >   I am a newbie to the asterisk and its architecture. :(
> > After reading some help in the tarball of Asterisk, I am
> > still in the mess. So I want to know where I can find a
> > detailed explanation of the Asterisk which including the
> > Architecture, Install, Configure, usage example document.
> > Maybe what I want is too much, after all it is a open
> > project, not commercial product. If I want to get that,
> > will I buy it or take participate in some course to learn
> > that ???
> 
> I found the same thing when I was first trying to get my head round it
> all, so once I had it worked out I setup a a-z on how to setup both
> linux (debian) and asterisk from cvs etc... http://www.asterisk.net.au
> 
> --
> 
> Best regards,
>   Duane
> 
> http://www.cacert.org - Free Security Certificates
> http://www.nodedb.com - Think globally, network locally
> http://www.sydneywireless.com - Telecommunications Freedom
> http://happysnapper.com.au - Sell your photos over the net!
> http://e164.org - Using Enum.164 to interconnect asterisk servers
> 
> "In the long run the pessimist may be proved right,
>  but the optimist has a better time on the trip."
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Re: [Asterisk-Users] Slackware + Asterisk + asterisk-addons

2005-01-30 Thread Mike Machado


There were some changes recently to the internal structure of the
realtime config. Make sure you have the latest CVS copy of both asterisk
and asterisk-addons and it should fix the compile error with
res_config_mysql.


On Sun, 2005-01-30 at 23:16 -0500, Bobby Lacey wrote:
> Hello
>  
> I am trying to get asterisk-addons installed so that I can use the
> mysql cdr feature. OK, I have the MySQL server (mysqld) installed, but
> I noticed that mysql-devel is also required. I tried to compile
> asterisk-addons and got a:
>  
> --CUT---
> res_config_mysql.c:422: error: unknown field `realtime_multi_func'
> specified in initializer
> res_config_mysql.c:422: warning: excess elements in struct initializer
> res_config_mysql.c:422: warning: (near initialization for
> `mysql_engine')
> res_config_mysql.c:423: error: unknown field `update_func' specified
> in initializer
> res_config_mysql.c:424: warning: excess elements in struct initializer
> res_config_mysql.c:424: warning: (near initialization for
> `mysql_engine')
> res_config_mysql.c: In function `parse_config':
> res_config_mysql.c:491: warning: assignment makes pointer from integer
> without a cast
> /usr/include/asterisk/utils.h: At top level:
> res_config_mysql.c:418: error: storage size of `mysql_engine' isn't
> known
> make: *** [res_config_mysql.o] Error 1
> [EMAIL PROTECTED]:/usr/src/asterisk-addons#
> 
>  
> So I'm assuming that I need mysql-devel? I went to download the
> package from mysql.com, but the only format I saw was .rpm. I am using
> Slackware 10.0 so rpm doesnt want to work well. I get the following:
>  
> rpm -Uvh MySQL-devel-4.1.9-0.i386.rpm
> warning: MySQL-devel-4.1.9-0.i386.rpm: V3 DSA signature: NOKEY, key ID
> 5072e1f5
> error: Failed dependencies:
> /bin/sh is needed by MySQL-devel-4.1.9-0
>  
>  
>  Does anyone have any suggestions or help they could offer?
>  
> Thanks in advance..
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Re: [Asterisk-Users] CAC Access Bank

2005-01-30 Thread Matt Riddell

Matt Schulte wrote:
We have an old CAC and we're trying to get groundstart working on it, we
think it may be a dip switch setting. Does anyone have the config
settings and/or the manual to config this thing? Any help is greatly
appreciated :-)
I have a CAC Access Bank 1 with 24 fxs configured as follows:
The first blue dip switch has an arrow pointing towards an R.
so, from that end I have:
1,2,4,5,7,8,9,10 away from R and the rest towards the R.
:)
The big switches are all to normal (although the test one that makes all 
the phones ring is sometimes fun).

My zaptel.conf for it is:
span=1,0,0,d4,ami
loadzone=nz
defaultzone=nz
fxoks=1-24
And it seems to work fine.  I did mine though by the markings on the 
actual case...

--
Cheers,
Matt Riddell
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[Asterisk-Users] Slackware + Asterisk + asterisk-addons

2005-01-30 Thread Bobby Lacey

Title: Message



Hello
 
I am trying to get 
asterisk-addons installed so that I can use the mysql cdr feature. OK, I have 
the MySQL server (mysqld) installed, but I noticed that mysql-devel is also 
required. I tried to compile asterisk-addons and got a:
 
--CUT---
res_config_mysql.c:422: error: unknown field `realtime_multi_func' 
specified in initializerres_config_mysql.c:422: warning: excess elements in 
struct initializerres_config_mysql.c:422: warning: (near initialization for 
`mysql_engine')res_config_mysql.c:423: error: unknown field `update_func' 
specified in initializerres_config_mysql.c:424: warning: excess elements in 
struct initializerres_config_mysql.c:424: warning: (near initialization for 
`mysql_engine')res_config_mysql.c: In function 
`parse_config':res_config_mysql.c:491: warning: assignment makes pointer 
from integer without a cast/usr/include/asterisk/utils.h: At top 
level:res_config_mysql.c:418: error: storage size of `mysql_engine' isn't 
knownmake: *** [res_config_mysql.o] Error 1[EMAIL PROTECTED]:/usr/src/asterisk-addons#
 
So I'm assuming that 
I need mysql-devel? I went to download the package from mysql.com, but the only 
format I saw was .rpm. I am using Slackware 10.0 so rpm doesnt want to work 
well. I get the following:
 
rpm -Uvh 
MySQL-devel-4.1.9-0.i386.rpmwarning: MySQL-devel-4.1.9-0.i386.rpm: V3 DSA 
signature: NOKEY, key ID 5072e1f5error: Failed 
dependencies:    /bin/sh is needed by 
MySQL-devel-4.1.9-0
 
 
 Does anyone 
have any suggestions or help they could offer?
 
Thanks in 
advance..
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Re: [Asterisk-Users] Japan

2005-01-30 Thread Jason Frisch

Sorry for my ignorance, but what is J1? I actually hope to use Softbanks 
fiber-based IPtel
service, but I believe they require VoIP TA so I guess the end result is 
just a standard
analog line.

Jason
Cory Andrews wrote:
Jason - I believe the Sangoma T1/E1/J1 boards may work in Japan, I 
will check and see if they offer driver support for J1, I do not know 
of an analog solution.

Cory Andrews
Senior Partner
VOIPSupply.com
+
800.398.VOIP X22
[EMAIL PROTECTED]

Jason Frisch wrote:
Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
finding hardware that is supported. I tried
Voicetronix but they said that they are too busy
to create a driver.
If anyone has had success with any particular
hardware please let me know. I plan to have
4-5 outside lines.
Thanks!
Jason
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Re: [Asterisk-Users] detailed asterisk howto

2005-01-30 Thread Duane

szj wrote:
Hi, all:
  I am a newbie to the asterisk and its architecture. :(
After reading some help in the tarball of Asterisk, I am
still in the mess. So I want to know where I can find a
detailed explanation of the Asterisk which including the
Architecture, Install, Configure, usage example document.
Maybe what I want is too much, after all it is a open
project, not commercial product. If I want to get that,
will I buy it or take participate in some course to learn
that ???
I found the same thing when I was first trying to get my head round it 
all, so once I had it worked out I setup a a-z on how to setup both 
linux (debian) and asterisk from cvs etc... http://www.asterisk.net.au

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
"In the long run the pessimist may be proved right,
but the optimist has a better time on the trip."
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Re: [Asterisk-Users] x100P wildcard discontinued ?

2005-01-30 Thread Cory Andrews

Varun - Model TDM01B is the replacement for the X100P, which has I 
believe been discontinued by Digium.

Cory Andrews
Senior Partner
VOIPSupply.com
+
800.398.VOIP X22
[EMAIL PROTECTED]

[EMAIL PROTECTED] wrote:
Hello,
 Is the x100P wildcard discontinued ?
If yes then which card replaces the x100P card ?
Thanks
Varun
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[Asterisk-Users] detailed asterisk howto

2005-01-30 Thread szj

Hi, all:
  I am a newbie to the asterisk and its architecture. :(
After reading some help in the tarball of Asterisk, I am
still in the mess. So I want to know where I can find a
detailed explanation of the Asterisk which including the
Architecture, Install, Configure, usage example document.
Maybe what I want is too much, after all it is a open
project, not commercial product. If I want to get that,
will I buy it or take participate in some course to learn
that ???
  Thanks for your advices.
  Best Regards.
  Sun Zongjun
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Re: [Asterisk-Users] Japan

2005-01-30 Thread Cory Andrews

Jason - I believe the Sangoma T1/E1/J1 boards may work in Japan, I will 
check and see if they offer driver support for J1, I do not know of an 
analog solution.

Cory Andrews
Senior Partner
VOIPSupply.com
+
800.398.VOIP X22
[EMAIL PROTECTED]

Jason Frisch wrote:
Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
finding hardware that is supported. I tried
Voicetronix but they said that they are too busy
to create a driver.
If anyone has had success with any particular
hardware please let me know. I plan to have
4-5 outside lines.
Thanks!
Jason
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Re: [Asterisk-Users] Asterisk@home and Zap Channels

2005-01-30 Thread Chuck Keeter

At 08:46 AM 1/30/2005, you wrote:
AMP does not support ZAP entensions. only sip and iax.
Maybe in a future release. You could post to the AMP
site and ask for this feature.
For now you will need to hack extensions.conf to get
it to work.
Thanks to everyone for the help, I have gotten it to answer inbound PSTN 
calls, and ring a SIP extension, I'm still threading my way through the 
extensions.conf and extensions_additional.conf to find where to add my ZAP 
channel extensions.

I have one more question that I can't seem to get straight, The ZAP channel 
phone, I can't dial any other extentions from it, I just get a fast busy. 
Same if I dial 9 to use the outside trunk. It works great from the SIP soft 
phone, but I can't seem to get the FXS phone to behave.

Can anyone point me in the correct direction?
Thanks again for all help..

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[Asterisk-Users] x100P wildcard discontinued ?

2005-01-30 Thread varun_saa

Hello,
  Is the x100P wildcard discontinued ?
If yes then which card replaces the x100P card ?

Thanks

Varun

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[Asterisk-Users] x100p issues + TDM400P

2005-01-30 Thread varun_saa

Hello,
  I have been wanting to use digium x100p
to get started. 

But it seems to have compatibility issues for different regions.
I am refering to the " 600 ohm US pstn standard only".

I am in India so If I were not to use x100p card then
what card I need to go in for?

I also read that x100p has been discontinued.

I am being recommended TDM400P wildcard. Is it OK
for India.

Does TDM400P wildcard has FXO and FXS ?

Thanks in advance

Varun

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[Asterisk-Users] Hitting IOCTL??

2005-01-30 Thread Robert Webb

I have recently started seeing the following message a lot: "We have hit
out IOCTL"

Can someone please explain what this means and/or how to fix it? It just
recently started appearing and seem to mainly come from when I hang up
an analog extension on a TDM400 card FXS port.

Robert



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[Asterisk-Users] Japan

2005-01-30 Thread Jason Frisch

Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
finding hardware that is supported. I tried
Voicetronix but they said that they are too busy
to create a driver.
If anyone has had success with any particular
hardware please let me know. I plan to have
4-5 outside lines.
Thanks!
Jason
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RE: [Asterisk-Users] where to buy x100p

2005-01-30 Thread Rich Adamson

I might also add some of the clone cards use an Intel chip. Those clones
that were sold in the US use one chip set, while those sold in some other
countries used a similar (but different number) chip set. Therefore,
some x100p clones _might_ work fine in non-600 ohm countries, but its
likely because that specific card was manufactured for non-US countries.

If you have one of those, check the Intel chip specs to see whether
you have the US or Non-US chip on the card.

In any case, the tdm card is probably a better bet for small 
installations. Let's just hope digium isolates the problem(s) soon.


> wow! excellent information, this should be added to the wiki under x100p / 
> tdm400.
> 
> Especially the info that the X100P is 600 ohm only would have made it a 
> lot clearer to me that I needn't buy an X100P (clone) for use in Europe
> 
> On Sun, 30 Jan 2005, Rich Adamson wrote:
> 
> >> Why does the X100P have echo and the Wildcard TDM400P have no echo?
> >>
> >> I thought the only advantage of using the TDM400P was that it used
> >> less interrupts than the X100P?
> >>
> >> Are there any other advantages?
> >
> > One of the major differences between the two cards is that tdm
> > fxo modules have support for multiple country pstn interfaces where
> > the x100p was designed for the 600 ohm US pstn standard only.
> > Mismatches associated with a 600 ohm card and pstn interface is
> > known to be a source of echo.
> >
> > The internal rtp data path in * for both the x100p and tdm card is
> > basically the same path (and the same echo canceller).
> >
> > The interrupts for both cards is exactly the same; 1,000 interrupts
> > per second. The interrupt rate is associated with the sampling of
> > digitized voice (data), and is not a hardware-card dependent item
> > in this case.
> >
> > The tdm card uses a Silcon Labs chipset that includes a two wire to
> > four wire hybrid, which _should_ contribute to reducing fxo echo.
> > However, the default driver does not currently preload (or dynamically)
> > adjust the hybrid coefficients to optimize the pstn line interface.
> > There has been some work going on recently to do that, but I don't
> > believe the code is production ready as yet. Setting the coefficients
> > is primarily a near-end echo optimization.
> >
> > The tdm card currently has another issue that causes at least some
> > implementations to fail after several days/weeks of operation. There
> > is some work going on at digium to identify the source of that problem,
> > however with intermitant problems that only show up over long periods
> > of time, it is very difficult to catch and identify the root cause.
> > The failure requires asterisk to be stopped and the card drivers to
> > be reloaded to correct the issue.
> >
> > Once the tdm issues have been resolved, its my opinion the card will
> > be a very cost effective way to interface to pstn lines in the small
> > office (and home) environment.
> >
> >
> >
> > ___
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---End of Original Message-


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Re: [Asterisk-Users] widcard x100P doubt

2005-01-30 Thread varun_saa



- Original Message -
From: Lyle Giese <[EMAIL PROTECTED]>
Date: Sunday, January 30, 2005 7:50 pm
Subject: Re: [Asterisk-Users] widcard x100P doubt

> Kewlstart is part of the spec for forward disconnect on loopstart 
> lines in
> North America(I am not familar with telco specs elsewhere in the 
> world).When the party at the other end of the call hangs up, the 
> telco switch
> removes talk battery from the circuit for a short period of time to 
> signaldisconnect.  This is also referred to as forward disconnect.
> 
> Lyle
> 
> Thanks Lyle

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Re: [Asterisk-Users] asterisk tries to dial out on lines already inuse.

2005-01-30 Thread Craig Guy

It sounds like you have multiple devices sharing the same physical lines?  I
think you will continue to have problems until you can rearrange the setup
to avoid line sharing to allow Asterisk to have dedicated access.  Might
have more luck with ISDN.

Craig

- Original Message - 
From: "Jon Gabrielson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Monday, January 31, 2005 8:14 AM
Subject: Re: [Asterisk-Users] asterisk tries to dial out on lines already
inuse.


> Asterisk should be able to do this, there are several cases
> when this is essential.  The first is a shared/party line where
> asterisk cannot have guaranteed access for whatever reason.
> In our case, that reason happens to be because we also use
> our outgoing lines for faxing.
> The second is that without dialtone detection, if for some
> reason the line is down, asterisk needs to know so that it can
> try a different outgoing line.  If the first line is down, asterisk
> shouldn't hang, it should wait a few seconds and try to dial
> out on the next line.
>
>
>
> Jon.
>
>
> On Sunday 30 January 2005 04:13 pm, Steven Critchfield wrote:
> > On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote:
> > > Can't asterisk look for a dialtone?  Even a $5 modem
> > > can detect whether or not there is a dialtone.
> >
> > Maybe you should just use your $5 modem and write your own software.
> >
> > Asterisk is a PBX. PBXs shouldn't have to deal with your bastardized
> > setup that doesn't respect the normal way in which a PBX is set up. A
> > PBX sits between the PSTN and ALL other access to the PSTN. In doing so,
> > asterisk can know ahead of time that the line is available. If you wait
> > for dialtone detection, then you have to also make code to understand
> > all international dialtones as well. Then you have to delay dial till
> > you are certain it is the tone you are expecting.
> >
> > > On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote:
> > > > > When I place a call with asterisk, asterisk will try to dial
> > > > > out on the first line even if the first line is already being
> > > > > used by someone else.  Any ideas on what I'm doing
> > > > > wrong?
> > > >
> > > > My question would be, how would asterisk know the line is in use if
it
> > > > isn't controlling it?
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Re: [Asterisk-Users] Processing incoming calls with multiple contextstover PRI

2005-01-30 Thread Lyle Giese




In zaptel.conf, put the line associated with 8350 
in the context bpns-external and when an external call comes in on 8350, it 
will drop to the s step in bpns-external.  I would suggest that you do 
something with the call if they don't bother to dial an extension, like send to 
a general voice mail box or ring all phones then drop int the general voice 
mail.
 
Lyle
 

  - Original Message - 
  From: 
  Jason Brown 
  To: asterisk-users@lists.digium.com 
  
  Sent: Sunday, January 30, 2005 7:59 
  PM
  Subject: [Asterisk-Users] Processing 
  incoming calls with multiple contextstover PRI
  
  
  So I have a problem. A customer of 
  mine wants a PBX, owns an office building. I want to sell him on 
  asterisk.  He has 4 tenants. I am using my asterisk box to simulate it. 
  My asterisk box has a TDM400P card, not a PRI card. Dont know if it makes any 
  difference.
   
  Anyway, I want to route incoming 
  phone calls to different contexts based on the phone number being 
  called.
   
  Here is my 
  extensions.conf
   
  [incoming-calls]
  exten => 
  _4078698350,1,Goto,bpns-external|${EXTEN}|1
  exten => 
  _4078698353,1,Goto,demo1-external|${EXTEN}|1
  exten => 
  _4078698359,1,Goto,demo2-external|${EXTEN}|1
  exten => 
  _4078698360,1,Goto,demo3-external|${EXTEN}|1
   
  [outgoing-calls]
  exten => 
  _407NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _321NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _1800NXX,1,DIal(ZAP/g1/${EXTEN},60)
  exten => 
  _1866NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _1877NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _1888NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},60)  
  ;voipjet NANPA
  exten => 
  _011.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},60) 
  ;voipjet WORLD
   
  [bpns-external]
  exten => 
  s,1,Playback,bpnsmenu
  exten => 
  1,1,Dial(SIP/1003,20,tr)
  exten => 
  1,2,Voicemail,u1003
  exten => 
  1,102,Voicemail,b1003
  exten => 
  2,1,Dial(SIP/1001,20,tr)
  exten => 
  2,2,Voicemail,u1001
  exten => 
  2,102,Voicemail,b1001
  exten => 
  3,1,Dial(SIP/1002,20,tr)
  exten => 
  3,2,VOicemail,u1002
  exten => 
  3,102,Voicemail,b1002
  exten => 
  1001,1,Dial(SIP/1001,20,tr)
  exten => 
  1001,2,Voicemail,u1001
  exten => 
  1001,102,VOicemail,b1002
  exten => 
  1002,1,Dial(SIP/1002,20,tr)
  exten => 
  1002,2,Voicemail,u1002
  exten => 
  1002,102,Voicemail,b1002
  exten => 
  1003,1,Dial(SIP/1003,20,tr)
  exten => 
  1003,2,Voicemail,u1003
  exten => 
  1003,102,Voicemail,b1003
  exten => 
  8500,1,VoicemailMain
  exten => 
  t,1,Hangup
   
  [bpns-internal]
  include => 
  outgoing-calls
  exten => 
  1001,1,Dial(SIP/1001,20,tr)
  exten => 
  1001,2,Voicemail,u1002
  exten => 
  1001,102,Voicemail,b1002
  exten => 
  1002,1,Dial(SIP/1002,20,tr)
  exten => 
  1002,2,Voicemail,u1002
  exten => 
  1002,102,Voicemail,u1002
  exten => 
  1003,1,Dial(SIP/1003,20,tr)
  exten => 
  1003,2,Voicemail,u1003
  exten => 
  1003,103,Voicemail,b1003
  exten => 
  1767,1,Dial(SIP/1001,20,tr)
  exten => 
  1767,2,Voicemail,u1001
  exten => 
  1767,102,Voicemail,b1001
  exten => 
  8500,1,VoicemailMain
   
  [demo1-external]
  exten => 
  s,1,Dial(SIP/1010,20,tr)
  exten => 
  s,2,Voicemail,u1010
  exten => 
  s,102,Voicemail,b1010
  exten => 
  8500,1,VoicemailMain
   
  [demo1-internal]
  include => 
  demo1-external
  include => 
  bpns-internal
  include => 
  outgoing-calls
   
  [demo2-external]
  exten => 
  s,1,Dial(SIP/1030,20,tr)
  exten => 
  s,2,Voicemail,u1030
  exten => 
  s,102,Voicemail,b1030
  exten => 
  8500,1,VoicemailMain
   
  [demo2-internal]
  include => 
  demo2-external
  include => 
  bpns-internal
  include => 
  outgoing-calls
   
  [demo3-external]
  exten => 
  s,1,Dial(SIP/2000,20,tr)
  exten => 
  s,2,Voicemail,u2000
  exten => 
  s,102,Voicemail,b2000
  exten => 
  8500,1,VoicemailMain
   
  [demo3-internal]
  include => 
  demo3-external
  include => 
  bpns-internal
  include => 
  outgoing-calls
   
  It doesnt work. I have a couple 
  asterisk guru friends who swear it should work. Here is what asterisk tells me 
  in verbose mode:
   
   
      -- Starting 
  simple switch on 'Zap/1-1'
  Jan 30 20:46:02 WARNING[7140]: 
  chan_zap.c:5586 ss_thread: CallerID returned with error on channel 
  'Zap/1-1'
    == Starting Zap/1-1 at 
  incoming-calls,s,1 failed so falling back to exten 
  's'
    == Starting Zap/1-1 at 
  incoming-calls,s,1 still failed so falling back to context 
  'default'
  Jan 30 20:46:02 WARNING[7140]: 
  pbx.c:1942 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in 
  context 'default', but no invalid handler
  n   
  Hungup 'Zap/1-1'  
  
   
  Now I understand it is looking for 
  the startup point. I dont understand why. 2 other asterisk guys I know swear 
  its supposed to work, although they are using sip/iax and not zap for 
  input.
   
  Anyone have any 
  ideas?
   
  Thanks
  
  

  ___

Re: [Asterisk-Users] Processing incoming calls with multiple contextstover PRI

2005-01-30 Thread david




Hi,Jason,
    The TDM400P card 
failed to get the Callee number or DID, so the * don't know how to route the 
call. There are something difference between the analog line and the PRI 
line.
 
    Regards.
 
    David
    http://www.iaxtalk.com
 

  - Original Message - 
  From: 
  Jason Brown 
  To: asterisk-users@lists.digium.com 
  
  Sent: Monday, January 31, 2005 9:59 AM
  Subject: [Asterisk-Users] Processing incoming 
  calls with multiple contextstover PRI
  
  
  So I have a problem. A customer of 
  mine wants a PBX, owns an office building. I want to sell him on 
  asterisk.  He has 4 tenants. I am using my asterisk box to simulate it. 
  My asterisk box has a TDM400P card, not a PRI card. Dont know if it makes any 
  difference.
   
  Anyway, I want to route incoming 
  phone calls to different contexts based on the phone number being 
  called.
   
  Here is my 
  extensions.conf
   
  [incoming-calls]
  exten => 
  _4078698350,1,Goto,bpns-external|${EXTEN}|1
  exten => 
  _4078698353,1,Goto,demo1-external|${EXTEN}|1
  exten => 
  _4078698359,1,Goto,demo2-external|${EXTEN}|1
  exten => 
  _4078698360,1,Goto,demo3-external|${EXTEN}|1
   
  [outgoing-calls]
  exten => 
  _407NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _321NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _1800NXX,1,DIal(ZAP/g1/${EXTEN},60)
  exten => 
  _1866NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _1877NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _1888NXX,1,Dial(ZAP/g1/${EXTEN},60)
  exten => 
  _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},60)  
  ;voipjet NANPA
  exten => 
  _011.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},60) 
  ;voipjet WORLD
   
  [bpns-external]
  exten => 
  s,1,Playback,bpnsmenu
  exten => 
  1,1,Dial(SIP/1003,20,tr)
  exten => 
  1,2,Voicemail,u1003
  exten => 
  1,102,Voicemail,b1003
  exten => 
  2,1,Dial(SIP/1001,20,tr)
  exten => 
  2,2,Voicemail,u1001
  exten => 
  2,102,Voicemail,b1001
  exten => 
  3,1,Dial(SIP/1002,20,tr)
  exten => 
  3,2,VOicemail,u1002
  exten => 
  3,102,Voicemail,b1002
  exten => 
  1001,1,Dial(SIP/1001,20,tr)
  exten => 
  1001,2,Voicemail,u1001
  exten => 
  1001,102,VOicemail,b1002
  exten => 
  1002,1,Dial(SIP/1002,20,tr)
  exten => 
  1002,2,Voicemail,u1002
  exten => 
  1002,102,Voicemail,b1002
  exten => 
  1003,1,Dial(SIP/1003,20,tr)
  exten => 
  1003,2,Voicemail,u1003
  exten => 
  1003,102,Voicemail,b1003
  exten => 
  8500,1,VoicemailMain
  exten => 
  t,1,Hangup
   
  [bpns-internal]
  include => 
  outgoing-calls
  exten => 
  1001,1,Dial(SIP/1001,20,tr)
  exten => 
  1001,2,Voicemail,u1002
  exten => 
  1001,102,Voicemail,b1002
  exten => 
  1002,1,Dial(SIP/1002,20,tr)
  exten => 
  1002,2,Voicemail,u1002
  exten => 
  1002,102,Voicemail,u1002
  exten => 
  1003,1,Dial(SIP/1003,20,tr)
  exten => 
  1003,2,Voicemail,u1003
  exten => 
  1003,103,Voicemail,b1003
  exten => 
  1767,1,Dial(SIP/1001,20,tr)
  exten => 
  1767,2,Voicemail,u1001
  exten => 
  1767,102,Voicemail,b1001
  exten => 
  8500,1,VoicemailMain
   
  [demo1-external]
  exten => 
  s,1,Dial(SIP/1010,20,tr)
  exten => 
  s,2,Voicemail,u1010
  exten => 
  s,102,Voicemail,b1010
  exten => 
  8500,1,VoicemailMain
   
  [demo1-internal]
  include => 
  demo1-external
  include => 
  bpns-internal
  include => 
  outgoing-calls
   
  [demo2-external]
  exten => 
  s,1,Dial(SIP/1030,20,tr)
  exten => 
  s,2,Voicemail,u1030
  exten => 
  s,102,Voicemail,b1030
  exten => 
  8500,1,VoicemailMain
   
  [demo2-internal]
  include => 
  demo2-external
  include => 
  bpns-internal
  include => 
  outgoing-calls
   
  [demo3-external]
  exten => 
  s,1,Dial(SIP/2000,20,tr)
  exten => 
  s,2,Voicemail,u2000
  exten => 
  s,102,Voicemail,b2000
  exten => 
  8500,1,VoicemailMain
   
  [demo3-internal]
  include => 
  demo3-external
  include => 
  bpns-internal
  include => 
  outgoing-calls
   
  It doesnt work. I have a couple 
  asterisk guru friends who swear it should work. Here is what asterisk tells me 
  in verbose mode:
   
   
      -- Starting 
  simple switch on 'Zap/1-1'
  Jan 30 20:46:02 WARNING[7140]: 
  chan_zap.c:5586 ss_thread: CallerID returned with error on channel 
  'Zap/1-1'
    == Starting Zap/1-1 at 
  incoming-calls,s,1 failed so falling back to exten 
  's'
    == Starting Zap/1-1 at 
  incoming-calls,s,1 still failed so falling back to context 
  'default'
  Jan 30 20:46:02 WARNING[7140]: 
  pbx.c:1942 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in 
  context 'default', but no invalid handler
  n   
  Hungup 'Zap/1-1'  
  
   
  Now I understand it is looking for 
  the startup point. I dont understand why. 2 other asterisk guys I know swear 
  its supposed to work, although they are using

Re: [Asterisk-Users] Processing incoming calls with multiple contextst over PRI

2005-01-30 Thread el Flynn

Jason Brown wrote:
So I have a problem. A customer of mine wants a PBX, owns an office
building. I want to sell him on asterisk.  He has 4 tenants. I am using
my asterisk box to simulate it. My asterisk box has a TDM400P card, not
a PRI card. Don't know if it makes any difference.

Just a guess about your problems, but if you have a PRI line incoming, wouldn't 
you need to connect it to a PRI card and not the TDM400P (which is for analog 
POTS lines)??

Flynn
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Re: [Asterisk-Users] Processing incoming calls with multiple contextst over PRI

2005-01-30 Thread Kevin P. Fleming

Jason Brown wrote:
Now I understand it is looking for the startup point. I don’t understand 
why. 2 other asterisk guys I know swear it’s supposed to work, although 
they are using sip/iax and not zap for input.
And why would you think those would act similarly? They don't.
Zap channels without ISDN or R2 signaling don't have a target extension 
to deliver to. They have no idea what phone number was called to make 
that channel ring, they only know the channel is ringing. They send the 
call to the "s" extension in whatever context you direct them to.

For now, you can simulate what you want by making each channel on the 
TDM400 go to a separate inbound context, and then using Goto() to go 
from there to the desired "incoming number" in your "incoming" context. 
With a PRI (which is what you would likely install in a real application 
of this type), you would actually receive the dialed number from the 
telco, and could route the call based on that.
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[Asterisk-Users] Meetme2 web - nothing happens on click ?

2005-01-30 Thread Robert Rozman

Hi,

I've installed meetme2 according to instructions. Everything seems ok,
members of conference are displayed, but nothing happens if I click on 3
action buttons (kick out, talk&listen, ...).

Any hints how to deal with this ?   In what way exactly does meetme2 kick
user off the conference ?

Thanks in advance,

regards,

Rob.

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[Asterisk-Users] Processing incoming calls with multiple contextst over PRI

2005-01-30 Thread Jason Brown









So I have a problem. A customer of mine wants a PBX, owns an
office building. I want to sell him on asterisk.  He has 4 tenants. I am
using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a
PRI card. Don’t know if it makes any difference.

 

Anyway, I want to route incoming phone calls to different
contexts based on the phone number being called.

 

Here is my extensions.conf

 

[incoming-calls]

exten => _4078698350,1,Goto,bpns-external|${EXTEN}|1

exten => _4078698353,1,Goto,demo1-external|${EXTEN}|1

exten => _4078698359,1,Goto,demo2-external|${EXTEN}|1

exten => _4078698360,1,Goto,demo3-external|${EXTEN}|1

 

[outgoing-calls]

exten => _407NXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _321NXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _1800NXX,1,DIal(ZAP/g1/${EXTEN},60)

exten => _1866NXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _1877NXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _1888NXX,1,Dial(ZAP/g1/${EXTEN},60)

exten =>
_1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},60) 
;voipjet NANPA

exten =>
_011.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},60)
;voipjet WORLD

 

[bpns-external]

exten => s,1,Playback,bpnsmenu

exten => 1,1,Dial(SIP/1003,20,tr)

exten => 1,2,Voicemail,u1003

exten => 1,102,Voicemail,b1003

exten => 2,1,Dial(SIP/1001,20,tr)

exten => 2,2,Voicemail,u1001

exten => 2,102,Voicemail,b1001

exten => 3,1,Dial(SIP/1002,20,tr)

exten => 3,2,VOicemail,u1002

exten => 3,102,Voicemail,b1002

exten => 1001,1,Dial(SIP/1001,20,tr)

exten => 1001,2,Voicemail,u1001

exten => 1001,102,VOicemail,b1002

exten => 1002,1,Dial(SIP/1002,20,tr)

exten => 1002,2,Voicemail,u1002

exten => 1002,102,Voicemail,b1002

exten => 1003,1,Dial(SIP/1003,20,tr)

exten => 1003,2,Voicemail,u1003

exten => 1003,102,Voicemail,b1003

exten => 8500,1,VoicemailMain

exten => t,1,Hangup

 

[bpns-internal]

include => outgoing-calls

exten => 1001,1,Dial(SIP/1001,20,tr)

exten => 1001,2,Voicemail,u1002

exten => 1001,102,Voicemail,b1002

exten => 1002,1,Dial(SIP/1002,20,tr)

exten => 1002,2,Voicemail,u1002

exten => 1002,102,Voicemail,u1002

exten => 1003,1,Dial(SIP/1003,20,tr)

exten => 1003,2,Voicemail,u1003

exten => 1003,103,Voicemail,b1003

exten => 1767,1,Dial(SIP/1001,20,tr)

exten => 1767,2,Voicemail,u1001

exten => 1767,102,Voicemail,b1001

exten => 8500,1,VoicemailMain

 

[demo1-external]

exten => s,1,Dial(SIP/1010,20,tr)

exten => s,2,Voicemail,u1010

exten => s,102,Voicemail,b1010

exten => 8500,1,VoicemailMain

 

[demo1-internal]

include => demo1-external

include => bpns-internal

include => outgoing-calls

 

[demo2-external]

exten => s,1,Dial(SIP/1030,20,tr)

exten => s,2,Voicemail,u1030

exten => s,102,Voicemail,b1030

exten => 8500,1,VoicemailMain

 

[demo2-internal]

include => demo2-external

include => bpns-internal

include => outgoing-calls

 

[demo3-external]

exten => s,1,Dial(SIP/2000,20,tr)

exten => s,2,Voicemail,u2000

exten => s,102,Voicemail,b2000

exten => 8500,1,VoicemailMain

 

[demo3-internal]

include => demo3-external

include => bpns-internal

include => outgoing-calls

 

It doesn’t work. I have a couple asterisk guru friends
who swear it should work. Here is what asterisk tells me in verbose mode:

 

 

    -- Starting simple switch on 'Zap/1-1'

Jan 30 20:46:02 WARNING[7140]: chan_zap.c:5586 ss_thread:
CallerID returned with error on channel 'Zap/1-1'

  == Starting Zap/1-1 at incoming-calls,s,1 failed so
falling back to exten 's'

  == Starting Zap/1-1 at incoming-calls,s,1 still
failed so falling back to context 'default'

Jan 30 20:46:02 WARNING[7140]: pbx.c:1942 ast_pbx_run:
Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no
invalid handler

n   Hungup
'Zap/1-1'  

 

Now I understand it is looking for the startup point. I don’t
understand why. 2 other asterisk guys I know swear it’s supposed to work,
although they are using sip/iax and not zap for input.

 

Anyone have any ideas?

 

Thanks






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Re: [Asterisk-Users] Caller ID in AU

2005-01-30 Thread Nathan Alberti

I have updated the Wiki with this info as I have seen it come up a few 
times.

Nathan.
Gary wrote:
Don't forget Howard, that Caller-ID presentation is an extra chargeable
service.
has it been turned on on these lines and confirmed ??
(its handy to carry a caller-id in your kit for checking:-)
On Sat, 29 Jan 2005 07:30:07 +1100, Howard Lowndes wrote:
 

On Fri, 2005-01-28 at 19:02, Simon Brown wrote:
   

Insert a Wait(2) before Answer
 

OK, I'll try that.  I have also done the suggested mod to the chan_zap.c
module to make the default rings 2.
   

Simon Brown 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Friday, 28 January 2005 17:30
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID in AU
Is anyone in AU successfully getting Caller ID from the analogue PSTN
service?
If so, what settings?
--
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux; when you want a
system that just works, you choose Microsoft."
--
"Flatter government, not fatter government; Get rid of the Australian
states."
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--
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."
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 http://lists.digium.com/mailman/listinfo/asterisk-users
   

.
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Re: [Asterisk-Users] Monitor calls timeout

2005-01-30 Thread el Flynn

jurgen wrote:

Problem is, Asterisk times out and disconnects after 10 seconds,
stopping the recording.
If I run something else in the context, say the infamous Monkey
Sounds, everything's fine, and the call just keeps going, annoying the
people on the line with monkey sounds. For some reason, the
*monitoring* always stops after 10 seconds.
did you try setting using AbsoluteTimeout in the context? e.g.
exten => s,1,Answer
exten => s,2,AbsoluteTimeout(0)
exten => s,3,Monitor(wav,testrecod,m)
I also once had a problem where my TDM400P card thought the far end had 
disconnected even though the two parties were still talking to each other. It 
was happening after roughly a minute and 40 seconds into the call.

Setting busydetect=no and callprogress=no in zapata.conf helped a bit, although 
I suspect it might actually had something to do with the phone line itself.

Flynn
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Re: [Asterisk-Users] New Firefly version

2005-01-30 Thread Duane

Adam Hart wrote:
Few people have claimed success, I'm not sure how though.
Any chance of a native linux version then? :)
--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
"In the long run the pessimist may be proved right,
but the optimist has a better time on the trip."
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Re: [Asterisk-Users] New Firefly version

2005-01-30 Thread Adam Hart

Duane wrote:
Adam Hart wrote:
As always, I'm happy to announce a new version of Firefly.
Firefly 1.9.8 has more of what you want and less of what you don't
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
There's a few bug fixes - notably fixed the Reject button and sending 
of audio before answering in some circumstances.

Has anyone been able to make firefly work under wine at all? If so how? 
A decent linux client is the only thing skype has over SIP/IAX...

Few people have claimed success, I'm not sure how though.
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Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

2005-01-30 Thread Jon Gabrielson

Asterisk should be able to do this, there are several cases
when this is essential.  The first is a shared/party line where 
asterisk cannot have guaranteed access for whatever reason.
In our case, that reason happens to be because we also use
our outgoing lines for faxing.
The second is that without dialtone detection, if for some 
reason the line is down, asterisk needs to know so that it can
try a different outgoing line.  If the first line is down, asterisk 
shouldn't hang, it should wait a few seconds and try to dial
out on the next line.



Jon.


On Sunday 30 January 2005 04:13 pm, Steven Critchfield wrote:
> On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote:
> > Can't asterisk look for a dialtone?  Even a $5 modem
> > can detect whether or not there is a dialtone.
>
> Maybe you should just use your $5 modem and write your own software.
>
> Asterisk is a PBX. PBXs shouldn't have to deal with your bastardized
> setup that doesn't respect the normal way in which a PBX is set up. A
> PBX sits between the PSTN and ALL other access to the PSTN. In doing so,
> asterisk can know ahead of time that the line is available. If you wait
> for dialtone detection, then you have to also make code to understand
> all international dialtones as well. Then you have to delay dial till
> you are certain it is the tone you are expecting.
>
> > On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote:
> > > > When I place a call with asterisk, asterisk will try to dial
> > > > out on the first line even if the first line is already being
> > > > used by someone else.  Any ideas on what I'm doing
> > > > wrong?
> > >
> > > My question would be, how would asterisk know the line is in use if it
> > > isn't controlling it?
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[Asterisk-Users] OH323 compile error : CVS-HEAD

2005-01-30 Thread M. Ehsanul Karim

I am getting the following error when compiling oh323-0.7.1 with
Asterisk CVS  (2004-12-21: Updated versions 0.7.1 (for Asterisk CVS
HEAD)

make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper'
make: *** [subdirs_build] Error 1
bash-2.05b# asteriskaudio.cxx:167: (Each undeclared identifier is
reported only once for
bash: syntax error near unexpected token `Each'


I have found other posting which says to use stable version but this
version of oh323 is supposed to wrok with Latest CVS-HEAD . Atleast
that is what told on the website of OH323.


Thanks.

Ehsanul Karim
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Re: [Asterisk-Users] Trying to make but it fails

2005-01-30 Thread Bob Goddard

On Sunday 30 January 2005 23:29, [EMAIL PROTECTED] wrote:
> I get these errors, and i am stuck here. I don't know what to do.
> I am using latest stable Zaptel 1.0.4 and Asterisk 1.0.5. I just ran it on
> a freinds machine and it went well there.
[...]
> chan_zap.c:3669: dereferencing pointer to incomplete type
> chan_zap.c:3670: confused by earlier errors, bailing out
> make[1]: *** [chan_zap.o] Error 2
> make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.5/channels'
> make: *** [subdirs] Error 1
> [EMAIL PROTECTED] asterisk-1.0.5]#

It's not the last errors which are important, but the first.


B
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[Asterisk-Users] Monitor calls timeout

2005-01-30 Thread jurgen

Hi all,

We're in a transition between OldPhoneSystem and Asterisk. One of the
things that's needed to be done right now with OldPhoneSystem is the
ability to record calls. I thought "Asterisk can record calls", so I
set about to make it happen. And it does, sort of.

I made a .call file that rings the exension that I want to have
recorded, and barges into the conversation, using a series of DTMF
codes that OldPhoneSystem understands. That bit works with no
problems. Once it's connected, the context I've placed the call into
looks like this:

exten => s,1,Answer
exten => s,2,Monitor(wav,testrecord,m)

And even that works - recording files are made called "testrecord"
that contain the conversation from the correct Zap channel.

Problem is, Asterisk times out and disconnects after 10 seconds,
stopping the recording.

If I run something else in the context, say the infamous Monkey
Sounds, everything's fine, and the call just keeps going, annoying the
people on the line with monkey sounds. For some reason, the
*monitoring* always stops after 10 seconds.

Here's what the console tells me:

-- Attempting call on Zap/4/442,55 for [EMAIL PROTECTED]:1 (Retry 1)
   > Channel Zap/4-1 was answered.
-- Executing Answer("Zap/4-1", "") in new stack
-- Executing SetVar("Zap/4-1",
"RECORDFILENAME=testrecording-s-20050131-102716") in new stack
-- Executing Monitor("Zap/4-1", "wav||m") in new stack

[all good so far]

Jan 31 10:27:26 WARNING[27937712]: pbx.c:1977 ast_pbx_run: Timeout,
but no rule 't' in context 'record'
-- Hungup 'Zap/4-1'

[okay, so I don't have a 't', but it shouldn't be timing out anyway!]

monitor executing ( nice -n 19 soxmix
//var/spool/asterisk/monitor/Zap-4-1-in.wav
//var/spool/asterisk/monitor/Zap-4-1-out.wav
//var/spool/asterisk/monitor/Zap-4-1.wav  && rm -f
//var/spool/asterisk/monitor/Zap-4-1-* ) &
Jan 31 10:27:26 NOTICE[27937712]: pbx_spool.c:244 attempt_thread: Call
completed to Zap/4/442,55

Does anyone have any ideas that could help here?

Thanks very much,

.jurgen


-- 
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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[Asterisk-Users] Trying to make but it fails

2005-01-30 Thread helpme

I get these errors, and i am stuck here. I don't know what to do.
I am using latest stable Zaptel 1.0.4 and Asterisk 1.0.5. I just ran it on a
freinds machine and it went well there.


chan_zap.c:3647: dereferencing pointer to incomplete type
chan_zap.c:3648: dereferencing pointer to incomplete type
chan_zap.c:3651: dereferencing pointer to incomplete type
chan_zap.c:3652: dereferencing pointer to incomplete type
chan_zap.c:3653: dereferencing pointer to incomplete type
chan_zap.c:3654: break statement not within loop or switch
chan_zap.c:3655: case label not within a switch statement
chan_zap.c:3656: case label not within a switch statement
chan_zap.c:3657: case label not within a switch statement
chan_zap.c:3658: case label not within a switch statement
chan_zap.c:3659: case label not within a switch statement
chan_zap.c:3660: dereferencing pointer to incomplete type
chan_zap.c:3661: break statement not within loop or switch
chan_zap.c:3662: default label not within a switch statement
chan_zap.c:3663: break statement not within loop or switch
chan_zap.c:3665: break statement not within loop or switch
chan_zap.c:3666: default label not within a switch statement
chan_zap.c:3667: dereferencing pointer to incomplete type
chan_zap.c:3669: dereferencing pointer to incomplete type
chan_zap.c:3670: confused by earlier errors, bailing out
make[1]: *** [chan_zap.o] Error 2
make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.5/channels'
make: *** [subdirs] Error 1
[EMAIL PROTECTED] asterisk-1.0.5]#


Per
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Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

2005-01-30 Thread Tim Mattison

Good call.

For our American readers... does anyone know where I can obtain a list
of states/counties and their regulations in regards to call recording?

On Sun, 2005-01-30 at 22:10 +, Mike Dent wrote:
> or maybe country? or should that be County? :)
> 
> Mike
> 
> 
> 
> On Sun, 30 Jan 2005 16:49:29 -0500, Tim Mattison <[EMAIL PROTECTED]> wrote:
> > Depending on your state.  :P
> > 
> > On Sun, 2005-01-30 at 11:42 -0500, Mark Phillips wrote:
> > > Don't forget to warn your callers about the recording.
> > >
> > > Tim Mattison wrote:
> > > > Try the monitor application instead of record.  I think that'll do what
> > > > you're looking for.
> > > >
> > > > On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote:
> > > >
> > > >>Hello All,
> > > >>
> > > >>I would like to record inbound and outbound calls to and from one
> > > >>number.
> > > >>
> > > >>I tried to add lines to my extensions.conf:
> > > >>
> > > >>DAY=`date "+%m-%d-%y_%H:%m"`
> > > >>
> > > >>;outbound
> > > >>exten => 551212,1,Record(${DAY}:gsm)
> > > >>exten => 551212,2,Dial(${TRUNKL3}/${EXTEN})
> > > >>
> > > >>;Inbound
> > > >>[line2]
> > > >>exten => 551212,1,Record(${DAY}:gsm)
> > > >>exten => 551212,2,Dial(SIP/101,20)
> > > >>exten => 551212,3,Hangup
> > > >>
> > > >>
> > > >>
> > > >
> > > >
> > > > ___
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> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] conference room capacity question

2005-01-30 Thread M.N.A.Smadi

hi;
have couple of questions regarding meet_me conference room application:
1) is there a maximum allowable number of concurrently active conference 
rooms per server?
2) what is the maximum allowable number of users in a given conference 
room before quailty creeps out?

thanks
moe smadi
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Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

2005-01-30 Thread Mike Dent

or maybe country? or should that be County? :)

Mike



On Sun, 30 Jan 2005 16:49:29 -0500, Tim Mattison <[EMAIL PROTECTED]> wrote:
> Depending on your state.  :P
> 
> On Sun, 2005-01-30 at 11:42 -0500, Mark Phillips wrote:
> > Don't forget to warn your callers about the recording.
> >
> > Tim Mattison wrote:
> > > Try the monitor application instead of record.  I think that'll do what
> > > you're looking for.
> > >
> > > On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote:
> > >
> > >>Hello All,
> > >>
> > >>I would like to record inbound and outbound calls to and from one
> > >>number.
> > >>
> > >>I tried to add lines to my extensions.conf:
> > >>
> > >>DAY=`date "+%m-%d-%y_%H:%m"`
> > >>
> > >>;outbound
> > >>exten => 551212,1,Record(${DAY}:gsm)
> > >>exten => 551212,2,Dial(${TRUNKL3}/${EXTEN})
> > >>
> > >>;Inbound
> > >>[line2]
> > >>exten => 551212,1,Record(${DAY}:gsm)
> > >>exten => 551212,2,Dial(SIP/101,20)
> > >>exten => 551212,3,Hangup
> > >>
> > >>
> > >>
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

2005-01-30 Thread Steven Critchfield

On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote:
> Can't asterisk look for a dialtone?  Even a $5 modem
> can detect whether or not there is a dialtone.

Maybe you should just use your $5 modem and write your own software. 

Asterisk is a PBX. PBXs shouldn't have to deal with your bastardized
setup that doesn't respect the normal way in which a PBX is set up. A
PBX sits between the PSTN and ALL other access to the PSTN. In doing so,
asterisk can know ahead of time that the line is available. If you wait
for dialtone detection, then you have to also make code to understand
all international dialtones as well. Then you have to delay dial till
you are certain it is the tone you are expecting.  

> On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote:
> > > When I place a call with asterisk, asterisk will try to dial
> > > out on the first line even if the first line is already being
> > > used by someone else.  Any ideas on what I'm doing
> > > wrong?
> >
> > My question would be, how would asterisk know the line is in use if it
> > isn't controlling it?
> >
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] where to buy x100p

2005-01-30 Thread Leo Ann Boon


The above assumes the x100p is connected to a pstn network that is
reasonably close to the 600 ohm US standard the card was designed to
interface with. Using the card in several other countries where the
pstn standards are different will likely result in echo that can not
be addressed with the above statements.
 

The x100p works fine in Singapore, probably because the terminal 
impedence here is 600 ohm as well.

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Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

2005-01-30 Thread Tim Mattison

Depending on your state.  :P

On Sun, 2005-01-30 at 11:42 -0500, Mark Phillips wrote:
> Don't forget to warn your callers about the recording.
> 
> Tim Mattison wrote:
> > Try the monitor application instead of record.  I think that'll do what
> > you're looking for.
> > 
> > On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote:
> > 
> >>Hello All,
> >>
> >>I would like to record inbound and outbound calls to and from one
> >>number.
> >>
> >>I tried to add lines to my extensions.conf:
> >>
> >>DAY=`date "+%m-%d-%y_%H:%m"`
> >>
> >>;outbound
> >>exten => 551212,1,Record(${DAY}:gsm)
> >>exten => 551212,2,Dial(${TRUNKL3}/${EXTEN})
> >>
> >>;Inbound
> >>[line2]
> >>exten => 551212,1,Record(${DAY}:gsm)
> >>exten => 551212,2,Dial(SIP/101,20)
> >>exten => 551212,3,Hangup
> >>
> >>
> >>
> > 
> > 
> > ___
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[Asterisk-Users] 302 Moved temporarily problem / Sipura 3000

2005-01-30 Thread Dan Fernandez




 
I can send calls from asterisk to a Sipura FXO 
interface (SIP/300) from any SIP phones including SIP/205 which is the Sipura 
3000 FXS interface. 
 
The problem I have is when a call from the PSTN is sends to Asterisk. On 
extnesion conf I dial all the SIP clients I get a 302 Moved temporarily 
when it dials SIP/205, the FXS interface. I have read on the bug tracker 
that ther is a patch with a new app SIPredirect (or similar) 
would this work for my problem. Any other thoughts?
 
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Re: [Asterisk-Users] Polycom changing policy - allowing firmwaredownloads?

2005-01-30 Thread Mark Eissler

On Jan 29, 2005, at 1:21 AM, Tim Courcy wrote:
Anyone can register on the PRC and get access to the Manuals. But you
have to be a certified reseller to get access to the firmware..

Gee, that would make me stay away from their phones
-mark
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[Asterisk-Users] Caller ID on H323

2005-01-30 Thread Krystian Filiks









Hi Friends

 

I have a problem presenting Caller ID on my H323 GW.

    

Scenario:  Sip
Phone à à Asterisk à à H323 GW à à PSTN (E1)

 

From PSTN to the Sip phone works fine I put this lines in extentions.conf

exten => 1234,1,SetCallerID,
“${CALLERIDNUM}”

exten => 1234,2,Dial(${testphone1},20,Ttm)

exten => 1234,3,Hangup

 

But from Sip to PSTN I get no CLI presentation. I tried the
following but no luck.

exten => _.,1,SetCallerID,
“${CALLERIDNUM}”

exten =>
_.,2,Dial(H323/${EXTEN})

exten => _.,3,Hangup

 

 

Can someone give me a pointer and maybe some config samples?

 

Thanks

KF






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RE: [Asterisk-Users] Digium and Intel Chipset compatability

2005-01-30 Thread gsr

I spent several days last year trying to get a TE410 to work on an HP DL360, 
but never seemed to get it to work.

I called Digium's (usually great) tech support at the time, and their response 
to me was 'if you can't get it to work we can't help you'. 

Please let me know if you have any success, I have a couple of extra dl360's 
in-stock that I would love to use as gateways...

-GSR

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: 27 January 2005 22:31
To: Martijn van Oosterhout; Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [Asterisk-Users] Digium and Intel Chipset compatability

We're having problems with a HP DL360 G4. TE410p simply does not 
generate any interrupts.

Digium tech support say that they will have some more information for me 
  by the end of the week.

Julian.

Martijn van Oosterhout wrote:
> Hi,
> 
> I'm going to be setting up some machines with 4 port E1 cards from
> Digium and I'm being told that TE410 is incompatable with several Intel
> chipsets including the ones in a lot of Dell server systems.
> 
> Is this true? I can't find any confirmed details on the mailing list
> about it. Also, the email seems to imply that the TE405P will be fine,
> though it doesn't say that explicitly.
> 
> Basically, is anyone using a 4 port E1 card successfully on an Intelï
> E7221 Chipset or similar?
> 
> Thanks in advance,

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[Asterisk-Users] One way call when the * server and phone in a local network

2005-01-30 Thread Dan Zhou

Hi everyone,
I started playing with Asterisk server a few days ago. So far, I only have 
made it 50% work.
Here is my situation, IP phone A and Asterisk server are in a same local 
network behind an ADSL router (public ip = a.b.c.d), and the * server is set 
as the DMZ host of the router.
IP phone B is in another network.

 IP phone A
 (10916) ---> ADSL router<-->internet<--> Rotuer2<--IP phone B
 asterisk server a.b.c.d   (10920)
 (192.168.1.2, DMZ host)
I can place call from phone A(10916) to B(10920) without any trouble.
But I cannot make a phone call from B to A.
I tried  "sip debug peer 10920" to check the SIP message flow, and found a 
problem.
Here is part of the log:

.
From: "10920" ;tag=zrgrUBMCJ7bVPtqC
To: "10916" 

SIP/2.0 404 Not Found

It figures out a wrong target sip address "10916" " 
for some reason. "sip:[EMAIL PROTECTED]" would be better?

When place call from 10916 to 10920, it looks fine:
..
From: "10916" ;tag=vmu4DFbjvCRUAuD8
To: "10920" 
..
I reckon because the * server is the DMZ host, which may affect the traffic 
to phone A that sits
in the same network. Is that the case? Any advice would be grealty 
appreciated! Thanks.

Cheers,
Dan
p.s my sip.conf
[general]
context=default		; Default context for incoming calls
port=5060			; UDP Port to bind to (SIP standard port is 5060)
bindaddr=192.168.1.2		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes			; Enable DNS SRV lookups on outbound calls
tos=0x18
disallow=all
allow=g729
allow=gsm
allow=ulaw
allow=alaw
allow=all
;If one is willing to forward some ports to the machine running asterisk, 
adding the following
;lines to the [general] section of sip.conf has been known to work;
nat=yes
qualify=yes

externip=a.b.c.d
localnet=192.168.1.0/255.255.255.0
;canreinvite=no
;the IP phone at home
[10916]
type=friend
secret=lucy
host=dynamic
callerid=Lucy <10916>
disallow=all
allow=g729
allow=ulaw
allow=alaw
nat=yes
qualify=yes
canreinvite=no
;the IP phone in friend's home
[10920]
type=friend
secret=dan
host=dynamic
callerid=Dan2 <10920>
disallow=all
allow=g729
allow=ulaw
allow=alaw
;allow=gsm
nat=yes
qualify=yes
canreinvite=no
_
FREE pop-up blocking with the new MSN Toolbar  get it now! 
http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

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Re: [Asterisk-Users] International Order for Grandstream and Sipura

2005-01-30 Thread Antonio Brandão

Dhennys ,

Estou interessado no SIPURA também. Você já tem alguma idéia em como adquirir?

Estou em São Carlos, SP. Podemos, eventualmente, dividir custos de envio.

-- 
Antonio José dos Santos Brandão



On Sat, 29 Jan 2005 05:20:30 -0200, Dhennys Pestana <[EMAIL PROTECTED]> wrote:
> (I appologize if this is off-topic, but I don't know where else to go.)
> 
> I'm interested in buying a few Grandstreams (HT486) and Sipuras (SPA2100) in
> bulk (maybe three units for a start, then a few dozens in a near future), but
> every company I find on the Web only ship using Fedex or UPS (at least US$80 
> per
> unit), which would increase the price to the sky since import taxes are
> calculated over the final price (product+shipping * taxes).
> 
> USPS, for instance, would charge only US$10 (with insurance) for each device.
> 
> Goods should be delivered to Brazil and payment must be made by international
> credit card or PayPal (confirmed address).
> 
> If there's ANYONE here interested in selling these devices (doens't matter if
> it's retail, used, OEM, open box, etc.), please get in touch with me in 
> private.
> 
> Thanks in advance,
> 
> -Dhennys Pestana
> Pestana Networks, Inc.
> 
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RE: [Asterisk-Users] where to buy x100p

2005-01-30 Thread Philipp von Klitzing

Hi!

> wow! excellent information, this should be added to the wiki under
> x100p / tdm400. 

So did _you_ add it?

Cheers, Philipp


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Re: [Asterisk-Users] Single or Dual Processor? High volume MeetMe

2005-01-30 Thread Philipp von Klitzing

Hi!

> Has anyone benchmarked Asterisk on a dedicated single versus dual 
> processor machine?

http://www.astertest.com/

Cheers, Philipp


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[Asterisk-Users] Callgroup with bristuff ISDN?

2005-01-30 Thread Remco Barende

Hi list!
I'm still trying to figure out about the groups in asterisk.
If I understand correctly, if you assign a certain group number and you 
assign the same call group number to a sip device the device will reing 
even though you did not specifically specify it in extension.conf?

How will this work for ISDN BRI/PRI?
I don't want some extensions to get all calls from the BRI/PRI, just the 
calls from one DID.

The wiki gives an example whereby a callgroup= is linked to a channel but 
this seems kinda silly with ISDN.

Can I use callgroups in such a setup, any config examples?
Thanks!!
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[Asterisk-Users] IAX2 firmware for PA168x (Giptel G100, Siptronic ST-100 etc)

2005-01-30 Thread Philipp von Klitzing

Hi there,

this is just a short note about one of the PA168x based phones out there 
which I obtained as "Giptel G100" (aka Siptronic ST-100): For some reason 
this phone would refuse to register with Asterisk using SIP, but after 
uploading the IAX2 firmware instead it finally came to life:

http://www.voip-info.org/tiki-index.php?page=GIPTEL+IP+Phones

Cheers, Philipp


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Re: [Asterisk-Users] Caller ID spoofing

2005-01-30 Thread Roger Schreiter

Calin Serbanescu schrieb:
unfortunatelly i have to accept their terms and rewrite caller id... but
again, i am newbie in scripting with agi and i can't find any example on
the net about this... do you have any link to such script ?

... what I should maybe also mention:
My script in the recent email is very general, not for
ISDN.
Even if you are beginner in scripting, you should read
the basics about AGI programming!
In the special case of ISDN, you'll need alternative
ways to set the callerid on the ISDN line - depending
on the hardare and driver you are using.
I can only tell for chan_capi:
Put in the dialstring in extensions.conf something like
CAPI/${CALLERID}:b${EXTEN}
or
CAPI/${new_callerid}:b${EXTEN}
In the latter case, your AGI script needs not to manipulate
the channel variable CALLERID, but your new variable
new_callerid with:
$AGI->set_variable("new_callerid", $new_callerid);
Hope, that helps!
Roger.
P.S.
Read the docs about your ISDN channel driver, e.g. chan_capi!
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Re: [Asterisk-Users] Caller ID spoofing

2005-01-30 Thread Calin Serbanescu

thanks very much, i'll try that... that is, no sleep until it's ready :)

On Sun, 2005-01-30 at 21:17 +0100, Roger Schreiter wrote:
> Calin Serbanescu schrieb:
> > ...
> > the net about this... do you have any link to such script ?
> 
> 
> No, I don't have such a link.
> But on the voip-wiki pages there are some examples
> for agi-scipting.
> 
> There are APIs for some common languages, e.g. Perl,
> which is maybe one of the fastet ways to code simple scripts.
> 
> Consult the README files on how to install the Perl module
> Asterisk::AGI.
> 
> Then try a little bit with:
> 
> 
> vvv
> #!/usr/bin/perl
> 
> use Asterisk::AGI;
> 
> $AGI = new Asterisk::AGI;
> 
> my %input = $AGI->ReadParse();
> $exten = $input{'extension'};
> $old_callerid = $input{'callerid'};
> 
> ... (some code to manipulte the callerid)
> 
> $AGI->set_callerid($new_callerid);
> 
> 
> 
> Roger.
> 
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Re: [Asterisk-Users] Caller ID spoofing

2005-01-30 Thread Roger Schreiter

Calin Serbanescu schrieb:
...
the net about this... do you have any link to such script ?

No, I don't have such a link.
But on the voip-wiki pages there are some examples
for agi-scipting.
There are APIs for some common languages, e.g. Perl,
which is maybe one of the fastet ways to code simple scripts.
Consult the README files on how to install the Perl module
Asterisk::AGI.
Then try a little bit with:
vvv
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
$exten = $input{'extension'};
$old_callerid = $input{'callerid'};
... (some code to manipulte the callerid)
$AGI->set_callerid($new_callerid);

Roger.
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Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

2005-01-30 Thread Paradise Dove

i have the same problem...
i've also added a feature request to bug tracker
(http://bugs.digium.com/bug_view_page.php?bug_id=0002612) regarding
this issue.



On Sun, 30 Jan 2005 13:40:06 -0600, Jon Gabrielson
<[EMAIL PROTECTED]> wrote:
> Can't asterisk look for a dialtone?  Even a $5 modem
> can detect whether or not there is a dialtone.
> 
> Thanks,
> 
> 
> Jon.
> 
> 
> On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote:
> > > When I place a call with asterisk, asterisk will try to dial
> > > out on the first line even if the first line is already being
> > > used by someone else.  Any ideas on what I'm doing
> > > wrong?
> >
> > My question would be, how would asterisk know the line is in use if it
> > isn't controlling it?
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Re: [Asterisk-Users] Caller ID spoofing

2005-01-30 Thread Calin Serbanescu

unfortunatelly i have to accept their terms and rewrite caller id... but
again, i am newbie in scripting with agi and i can't find any example on
the net about this... do you have any link to such script ?

thanks

On Sun, 2005-01-30 at 20:42 +0100, Roger Schreiter wrote:
> Calin Serbanescu schrieb:
> > ...
> > e164 numbers in my h.323 network and my ISDN provider doesn't accept
> > those identities (CIDs). So, i have to spoof the outgoing CID depending
> > on incoming CID. 
> > 
> > Is there any possible way of doing this by AGI? How? any examples are
> > welcome.
> 
> 
> Hi,
> 
> I'm not sure, whether I've really understood your desire.
> 
> I assume romanian ISDN is similar to the german one, and
> BRI or PRI connects are bound to a fixed set of numbers.
> 
> Manipulation of callerids when forwading calls from H.323
> to ISDN or vice versa, taking care of the above mentionned
> restrictions, is easily done with an AGI-script and
> maybe a (mySQL) database.
> 
> Alternatively look for a provider able to accept H.323 and
> passing callerids unchanged to PSTN. Of course that provider
> will bind you to some regulations in order not to abuse
> this feature to spoof callerids!
> 
> 
> Roger.
> 
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Re: [Asterisk-Users] Caller ID spoofing

2005-01-30 Thread Roger Schreiter

Calin Serbanescu schrieb:
...
e164 numbers in my h.323 network and my ISDN provider doesn't accept
those identities (CIDs). So, i have to spoof the outgoing CID depending
on incoming CID. 

Is there any possible way of doing this by AGI? How? any examples are
welcome.

Hi,
I'm not sure, whether I've really understood your desire.
I assume romanian ISDN is similar to the german one, and
BRI or PRI connects are bound to a fixed set of numbers.
Manipulation of callerids when forwading calls from H.323
to ISDN or vice versa, taking care of the above mentionned
restrictions, is easily done with an AGI-script and
maybe a (mySQL) database.
Alternatively look for a provider able to accept H.323 and
passing callerids unchanged to PSTN. Of course that provider
will bind you to some regulations in order not to abuse
this feature to spoof callerids!
Roger.
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Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

2005-01-30 Thread Jon Gabrielson

Can't asterisk look for a dialtone?  Even a $5 modem
can detect whether or not there is a dialtone.

Thanks,


Jon.


On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote:
> > When I place a call with asterisk, asterisk will try to dial
> > out on the first line even if the first line is already being
> > used by someone else.  Any ideas on what I'm doing
> > wrong?
>
> My question would be, how would asterisk know the line is in use if it
> isn't controlling it?
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[Asterisk-Users] Single or Dual Processor? High volume MeetMe

2005-01-30 Thread Spencer Nassar

Has anyone benchmarked Asterisk on a dedicated single versus dual 
processor machine?  Or could any Asterisk developers comment on whether 
it is architected in such a way that threads could run on multiple CPUs 
(especially MeetMe2)?

At a higher level, can I host more simultaneous lines and/or 
conferences for MeetMe if I use a dual processor machine versus single? 
 Also, any info on memory use with high numbers of conference users 
(100, 1000)?

Thanks!
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Re: [Asterisk-Users] Vocera Badges

2005-01-30 Thread Andrew Thompson

John Middleton wrote:
Anyone got any experiences of these with *, and also costings?
Someone mentioned them on the list several months ago, but I don't think 
anyone mentioned actually using it.

--
Andrew Thompson
http://aktzero.com/
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[Asterisk-Users] Caller ID spoofing

2005-01-30 Thread Calin Serbanescu

Hello everybody!

I am having the following problem and since I am a beginner in playing
with asterisk, i can't solve it:

I am trying to  integrate my existing H.323 network in real world
telephony by ISDN cards. The problem is that i DON'T want to change all
e164 numbers in my h.323 network and my ISDN provider doesn't accept
those identities (CIDs). So, i have to spoof the outgoing CID depending
on incoming CID. 

Is there any possible way of doing this by AGI? How? any examples are
welcome.

Also, if i don't ask too much, it would be great to integrate this "CID
translation" table in my existing MySQL database for Asterisk.

Thanks, 
Calin Serbanescu

IT Manager
TransTel Services

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Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

2005-01-30 Thread Andrew Kohlsmith

On January 30, 2005 12:18 pm, Brian Dingman wrote:
> There is the little problem of having to switch numbers and then
> communicating to everyone that the number has changed. This also only
> seems to be a problem on inbound calls.

And why, praytell, did you go into production with DIDs from a provider that 
couldn't be ported and without adequate testing?

I could be wrong and maybe it worked all along until this point, but in that 
case you should be able to revert to the last good config and continue.

-A.
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RE: [Asterisk-Users] Asterisk on MS Virtual Server

2005-01-30 Thread Bill Seddon

<< Were the pops/drops/buzzes a problem only with communications via a
telephony card?>>

We ran Asterisk in a VPC instance (Redhat 8.0) for 3 months while we
evaluated Asterisk.  The only reason we had to move to a version of Linux
running directly on hardware was a need to run X101P cards.

We had no sound issues except when the host machine was printing.  The host
ran (still does) Windows 2000 Server, has an AMD Duron processor and 1GB
RAM.

Bill Seddon

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Courtnage
Sent: January 30, 2005 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk on MS Virtual Server

Hi Greg,

On Sun, 2005-30-01 at 09:42 -0500, Greg Boehnlein wrote:
> It "works", but you will have timing issues and very poor audio quality.
> 
> I've run linux both under Vmware as well as running it under CoLinux 
> directly on windows w/ no emulation neccessary. All emulation / 
> virtualization layers have a problem whereby they are not able to keep up 
> with the interrupt frequency that good quality audio requires, so you will

> hear pops, drops and buzzes in your conversation.

Were the pops/drops/buzzes a problem only with communications via a
telephony card?  I'm curious to know if one could run * in a vmware
session (only VoIP trunks), and if a telephony card is required, run
another instance of * (and the card drivers) on the host OS.

Thanks
Ryan

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[Asterisk-Users] Asterisk and Grandstreams on marginal broadband...

2005-01-30 Thread Kim Lux

We've got an office with a marginal broadband connection.

Do you think Asterisk + Grandstreams can be tuned to give a good quality
call on with this jitter/latency ?  This is the only broadband supplier
in the area, so changing carriers would be difficult.

I made a call from the Grandstream with * and it was a bit choppy and
had some echo.


Thanks. 

ping sip.babytel.ca
PING sip.babytel.ca (216.18.125.3) 56(84) bytes of data.
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=0 ttl=51
time=64.9 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=1 ttl=51
time=76.0 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=2 ttl=51
time=150 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=3 ttl=51
time=130 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=4 ttl=51
time=87.9 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=5 ttl=51
time=82.2 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=6 ttl=51
time=128 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=7 ttl=51
time=174 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=8 ttl=51
time=68.6 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=9 ttl=51
time=125 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=10
ttl=51 time=154 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=11
ttl=51 time=207 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=12
ttl=51 time=115 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=13
ttl=51 time=107 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=14
ttl=51 time=97.4 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=15
ttl=51 time=76.3 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=16
ttl=51 time=119 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=17
ttl=51 time=122 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=18
ttl=51 time=72.4 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=19
ttl=51 time=106 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=20
ttl=51 time=109 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=21
ttl=51 time=78.4 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=22
ttl=51 time=130 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=23
ttl=51 time=153 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=24
ttl=51 time=200 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=25
ttl=51 time=229 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=26
ttl=51 time=146 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=28
ttl=51 time=153 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=29
ttl=51 time=140 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=30
ttl=51 time=90.9 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=31
ttl=51 time=69.4 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=32
ttl=51 time=159 ms
64 bytes from sprox-001.mtl.pop.vds.ca (216.18.125.3): icmp_seq=33
ttl=51 time=71.5 ms

--- sip.babytel.ca ping statistics ---
34 packets transmitted, 33 received, 2% packet loss, time 33024ms
rtt min/avg/max/mdev = 64.993/121.282/229.137/42.306 ms, pipe 2



-- 
Kim Lux,  Diesel Research Inc.


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Re: [Asterisk-Users] iaxComm version 1.0 released

2005-01-30 Thread rsenykoff






>Any work to support some USB Phones!? The ability to dial using the phones

>keypad!?

Not yet, but I'll probably add suport for the TigerJet phone eventually.


That would be -awesome.- The mgmt at my company wants that ability, and
integrating with IAX protocol would make portability huge.

-Ron

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Re: [Asterisk-Users] Asterisk on MS Virtual Server

2005-01-30 Thread Ryan Courtnage

Hi Greg,

On Sun, 2005-30-01 at 09:42 -0500, Greg Boehnlein wrote:
> It "works", but you will have timing issues and very poor audio quality.
> 
> I've run linux both under Vmware as well as running it under CoLinux 
> directly on windows w/ no emulation neccessary. All emulation / 
> virtualization layers have a problem whereby they are not able to keep up 
> with the interrupt frequency that good quality audio requires, so you will 
> hear pops, drops and buzzes in your conversation.

Were the pops/drops/buzzes a problem only with communications via a
telephony card?  I'm curious to know if one could run * in a vmware
session (only VoIP trunks), and if a telephony card is required, run
another instance of * (and the card drivers) on the host OS.

Thanks
Ryan

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Re: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch]

2005-01-30 Thread Olle E. Johansson

Geoff Speicher wrote:
Sipura has implemented auto-answer in version 0.9.5 of the SPA-841
firmware.  However, it is implemented via the Call-Info header, which
Asterisk stable doesn't currently support.
The attached patch implments a quick hack to support the Call-Info
header from the Dial() application by way of setting the CALL_INFO
variable.  For example, the following macro can be used to dial up a
single extension as an auto-answer intercom, or multiple extensions as a
system-wide page.  Note that the SPA-841 requires the leading semicolon
for auto-answer to work.
This was exactly the reason I wrote the sipaddheader() function...
We can't go on adding extra code for every manufacturer's extra 
non-standard headers.

/Olle
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Re: [Asterisk-Users] Asterisk on MS Virtual Server

2005-01-30 Thread Greg Boehnlein

On Sun, 30 Jan 2005, Paul Tyreman wrote:

> http://www.digium.com/index.php?menu=astwind
> 
> I think this may be worth a look, I'm downloading it as I type this 
> e-mail...
> 
> I didn't know Asterisk had the possibility of being run on a windows machine 
> and while it's not as stable as a Linux implementation, it might just do for 
> the moment, as I don't have many users.
> 
> Is there any documentation on this windows based software, or if not, do you 
> know where I can get more info on it ?
> 
> Thanks, Paul.

Paul,
Since I maintain AstWind, please feel free to give me your 
feedback when you get a chance. I'm working on re-packaging and updating 
AstWind so that it contains a 2.6.8 Colinux kernel, 1.0.5 Asterisk and all 
of the latest Debian Updates, with a real installer. I.E. not a crappy 
batch file that just copies stuff over! ;)

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RE: [Asterisk-Users] where to buy x100p

2005-01-30 Thread Remco Barende

wow! excellent information, this should be added to the wiki under x100p / 
tdm400.

Especially the info that the X100P is 600 ohm only would have made it a 
lot clearer to me that I needn't buy an X100P (clone) for use in Europe

On Sun, 30 Jan 2005, Rich Adamson wrote:
Why does the X100P have echo and the Wildcard TDM400P have no echo?
I thought the only advantage of using the TDM400P was that it used
less interrupts than the X100P?
Are there any other advantages?
One of the major differences between the two cards is that tdm
fxo modules have support for multiple country pstn interfaces where
the x100p was designed for the 600 ohm US pstn standard only.
Mismatches associated with a 600 ohm card and pstn interface is
known to be a source of echo.
The internal rtp data path in * for both the x100p and tdm card is
basically the same path (and the same echo canceller).
The interrupts for both cards is exactly the same; 1,000 interrupts
per second. The interrupt rate is associated with the sampling of
digitized voice (data), and is not a hardware-card dependent item
in this case.
The tdm card uses a Silcon Labs chipset that includes a two wire to
four wire hybrid, which _should_ contribute to reducing fxo echo.
However, the default driver does not currently preload (or dynamically)
adjust the hybrid coefficients to optimize the pstn line interface.
There has been some work going on recently to do that, but I don't
believe the code is production ready as yet. Setting the coefficients
is primarily a near-end echo optimization.
The tdm card currently has another issue that causes at least some
implementations to fail after several days/weeks of operation. There
is some work going on at digium to identify the source of that problem,
however with intermitant problems that only show up over long periods
of time, it is very difficult to catch and identify the root cause.
The failure requires asterisk to be stopped and the card drivers to
be reloaded to correct the issue.
Once the tdm issues have been resolved, its my opinion the card will
be a very cost effective way to interface to pstn lines in the small
office (and home) environment.

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[Asterisk-Users] Re: Sipura SPA-841 auto-answer support [patch]

2005-01-30 Thread Geoff Speicher

> The best place for patches to Asterisk is the Asterisk bug tracker,
> which can be found at http://bugs.digium.com .
> 
> *Please* don't forget to read the guidelines:
> http://www.digium.com/bugguidelines.html and file a disclaimer.
> 
> Thanks for the patch! I'll look for it on the bug tracker :)

Actually, I didn't really expect that the patch would be committed to
Asterisk CVS, for a couple of reasons:

  1. Asterisk devel already has sipAddHeader (or something like that),
  which is a more generally useful way to accomplish the same thing.
  Until that function makes its way to a stable release, I just need a
  quick, stable way to support Call-Info, and I assumed that I wasn't
  the only one.

  2. It ain't the prettiest hack.  Granted it's simple, but it's making
  an existing problem worse.  Take a look at transmit_reinvite().  I'm
  reminded of a favorite fortune(6) of mine:

If you have a procedure with 10 parameters, you probably missed some.

  char* call_info is parameter number nine.  :)

  A better patch would create a more flexible mechanism for Dial to pass
  information from dialplan variables into transmit_reinvite().

That being said, if enough people find this patch useful, then I'd
certainly file whatever disclaimers necessary to get it committed.

Geoff

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[Asterisk-Users] Thank you

2005-01-30 Thread Jeff Konrade-Helm



Thanks to everyone who responded with both advice and words of caution. 

I have no doubt I am going to be getting in over my head. That is just my
M.O. but I'm still surviving an arguably better off for it. 

The bottom line is this: my organization needs a phone system. We can't
afford state-of-the-art, plug-n-play, proprietary (yuck!) and buying
something 10+ years old will likely leave us in the same boat the next time
we need to expand. 

I was very apprehensive but after receiving such an overwhelming welcome to
this group and offers of advice and support, I am certain this is the right
choice. 

Wish me luck and I'm sure you will be hearing from me. 

Thanks again. 

jeff
:-j
 
Jeff Konrade-Helm
Director of Communications and Marketing
Autism Society of Colorado
701 S. Logan St.; Suite 103
Denver, CO 80209-4169
720.214.0794
720.274.2744 - fax
[EMAIL PROTECTED]
www.autismcolorado.org



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5902 spam emails to date. Paying users do not have this message in their
emails. Try www.SPAMfighter.com for free now!

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[Asterisk-Users] D/41D

2005-01-30 Thread Nash, Jason

Hello,
I'm attempting to setup a test asterisk server.  I have a couple of old 4
port D/41D ISA dialogic cards.  Has anyone had any success at getting them
to work?  I've done some searching on the internet and it seems like some
have them working but I have not been able to find any help docs on how to
set it up, or which drivers to use.
Any help would be greatly appreciated!
Thanks
Jason
This message along with any attachments is intended only for the use of the
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that is confidential and prohibited from disclosure.  If you are not the
intended recipient, you are hereby notified that any dissemination or
copying of this message or attachment is strictly prohibited.  If you have
received this message in error, please notify the original sender
immediately by telephone or return e-mail and delete this message along with
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Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

2005-01-30 Thread Brian Dingman

There is the little problem of having to switch numbers and then
communicating to everyone that the number has changed. This also only
seems to be a problem on inbound calls.


On Sat, 29 Jan 2005 23:34:49 -0500, Andrew Kohlsmith
<[EMAIL PROTECTED]> wrote:
> On January 29, 2005 11:29 pm, Brian Dingman wrote:
> > This is driving me crazy. I have resorted to using the m option in the
> > Dial command just so folks don't hang up. I can't believe nobody else
> > is having this issue.
> 
> Simple test: try it with another VOIP provider.  Throw $5 at a nufone account,
> or an iax.cc account.  See what happens.  Hell you're already saying it's
> working with other providers, so what's your data showing you?
> 
> Why do people insist on staying with VOIP providers who provide spotty
> performance and half-assed answers to technical support issues?
> 
> -A.
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Re: [Asterisk-Users] Silly question: Why multiple lines on SIP phones?

2005-01-30 Thread Michael Graves

On Sat, 29 Jan 2005 21:17:56 -0500 (EST), Paul Dugas wrote:

>This is probably going to sound really silly and I must be confused about
>it.  Maybe someone can set me straight.
>
>I've been tinkering for a while with * and a number of different FXO/FXS
>cards, SIP phones, and ATAs trying to get a feel for what works and what
>doesn't.  In the SIP phone group, I have a Cisco 7940G, a Polycom 500, and
>now an SPA-841.  Each of these allow me to configure at least 2 "lines". 
>What I'm wrestling with is what to do with the second lines.  With the
>cisco and the polycom, I have only one line configured and can
>successfully conference and transfer calls.  When I'm on the phone and a
>new call comes in, they display the second call and controls for switching
>between them.  (hmmm... what happens when the third comes along?)  The
>Sipura appears to be unable to do this without configuring a second
>extension.  (anybody got any insight on that?)  My question is basically,
>assuming all I want is the same kind of service I get on a traditional
>analog line with respect to call waiting and 3-way, do I need a multi-line
>phone?
>
>Back to the Sipura... If I configure it to only register the first line,
>it seems it cannot conference or transfer calls.  With my current configs,
>a second incoming call to the extension dumps straight to "busy" voice
>mail.  I guess I could setup my dial plan to roll to the phone's line 2
>but that seems like a mess.  It's not needed with the cisco or polycom so
>it's possible, right?

Some phones don't support multiple lines at all. The Zultys 4x4 and 4x5
phones provide 4 "call appearances" but only one registration. That
means that the phone is one extension, one line. However, a second,
third or fourth call incomming/outgoing uses another call appearance.
This acts sort of like a local hunt group...and has turned out to be
very natural to use. There's less administrative and dialplan overhead
than with my Polycom phones.

I started out with three extensions programmed in my Polycom IP600
phones, but eventually knocked them back to only 1. With each extension
having two call appearancs on the IP600 it's more than a mere mortal
can do to manage all the possible call activity.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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[Asterisk-Users] Setting call forward for Agent's in a Queue

2005-01-30 Thread Wessel de Roode

Hi!,

I'm trying to set up a Queue (which works fine now :-)
Sip clients can login in to the Queue with dialing 91 on there phone.
And as soon as there are customers the Queue calls the agents back.
I would like that the queue calls the agents also if it's phone is
call-forwarded.

With agents (sip clients) are added with the following extensions:

exten => 91,1,AddQueueMember(myqueue)
exten => 91,2,Playback(agent-loginok)
exten => 91,3,Hangup

However if I use the following script to enable call-forwarding:

exten => _*21*X.#,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten => _*21*X.#,2,Answer
exten => _*21*X.#,3,Playback(call-fwd-unconditional,skip)
exten => _*21*X.#,4,Hangup

And the following macro for every internal dial command:

[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto 102
exten=s,2,Dial(Local/[EMAIL PROTECTED]/n)   ; Unconditional forward
exten=s,3,Dial(${ARG2},20) ; 20sec timeout
exten=s,4,DBget(temp=CFBS/${ARG1})  ; Get CFBS key, if not existing, goto
105
exten=s,5,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on busy or unavailable
exten=s,102,Goto(s,3)
exten=s,105,Busy

It is not working as the Queue is dialing directly the extension of the sip
phone.
Any alternatives or workarounds?

Many thanks in advance...

Wessel

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Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

2005-01-30 Thread Mark Phillips

Don't forget to warn your callers about the recording.
Tim Mattison wrote:
Try the monitor application instead of record.  I think that'll do what
you're looking for.
On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote:
Hello All,
I would like to record inbound and outbound calls to and from one
number.
I tried to add lines to my extensions.conf:
DAY=`date "+%m-%d-%y_%H:%m"`
;outbound
exten => 551212,1,Record(${DAY}:gsm)
exten => 551212,2,Dial(${TRUNKL3}/${EXTEN})
;Inbound
[line2]
exten => 551212,1,Record(${DAY}:gsm)
exten => 551212,2,Dial(SIP/101,20)
exten => 551212,3,Hangup


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Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

2005-01-30 Thread Wilson Pickett

> When I place a call with asterisk, asterisk will try to dial
> out on the first line even if the first line is already being
> used by someone else.  Any ideas on what I'm doing
> wrong?

My question would be, how would asterisk know the line is in use if it
isn't controlling it?
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[Asterisk-Users] Vservices.inv of Julian Pawlowski anoyne has the macro-dailer for this?

2005-01-30 Thread Wessel de Roode

Hi,

I've found the Vertical Service Codes / vservices.inc of Julian in the cache
of google.
It's an very extended extensions include with all the *21 *67 etc services
implemented so it is stored to ODBC or if you replace it to Dbget/put etc.

I'm wondering if somebody has the macro/agi for using these extensions once
stored in the Asterisk db or ODBC.

Or am I missing something? And will it work just as it is under ODBC


Wessel

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RE: [Asterisk-Users] agent logoff

2005-01-30 Thread Joe Dennick

I have a separate extension set up for logoff that doesn't pass the
callerid to AgentCallbacklogin.  So my agents dial 301 to log on and 302
to logoff.  A lot of the phones (Cisco for example) allow you set up
speed dials, so its even easier for an agent to log on and off.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Fernandez
Sent: Sunday, January 30, 2005 10:25 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] agent logoff



I am using AgentCallbacklogin to logon agents. I am trying to avoid
agents being logged in more than once in different extensions (is this a
bug?) by passing the callerid to the AgentCallbacklogin function as an
option. The problem is that by doing this, agents are not asked for an
extension and they cannot logoff (by pressing the #).

Any ideas how can agents logoff? 

-- 
Internal Virus Database is out-of-date.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 1/21/2005
 

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Re: [Asterisk-Users] Anyone having problems with LiveVoIP?

2005-01-30 Thread Tim Lewis

LiveVoIP has many problems. #1 seems to be customer service. #2 they use
Level 3. I am switching to Txlink

On Sun, 2005-01-30 at 09:15, Ryan Laginski wrote:
> Hi,
> I made the mistake of ordering an 800 number as well. I have the same
> problem you have, asterisk registers, but I get a fast busy when
> dialing the number. Checking the cdr, and it shows the call has been
> placed, but for 0.0 minutes.
> 
> I have yet to hear from them as well. Reading other posts, people say
> they are very responsive.
> -Ry
> 
> 
> On Thu, 27 Jan 2005 12:59:17 -0500, Glenn Powers <[EMAIL PROTECTED]> wrote:
> > 
> > I ordered an 800# from LiveVoIP two days ago. I can register with
> > Asterisk just fine, but when I call my 800#, I get a fast busy. I
> > emailed support a day and a half ago and have heard NOTHING from them.
> > 
> > VoicePulse Connect and VoipJet both work great for me.
> > 
> > Someone on -users said "you get what you pay for" regarding LiveVoip.
> > They couldn't have been more correct!
> > 
> > cheers,
> > glenn
> > 
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] where to buy x100p

2005-01-30 Thread Rich Adamson

> Why does the X100P have echo and the Wildcard TDM400P have no echo?
> 
> I thought the only advantage of using the TDM400P was that it used 
> less interrupts than the X100P?
> 
> Are there any other advantages?

One of the major differences between the two cards is that tdm
fxo modules have support for multiple country pstn interfaces where
the x100p was designed for the 600 ohm US pstn standard only. 
Mismatches associated with a 600 ohm card and pstn interface is 
known to be a source of echo.

The internal rtp data path in * for both the x100p and tdm card is 
basically the same path (and the same echo canceller).

The interrupts for both cards is exactly the same; 1,000 interrupts
per second. The interrupt rate is associated with the sampling of
digitized voice (data), and is not a hardware-card dependent item
in this case.

The tdm card uses a Silcon Labs chipset that includes a two wire to
four wire hybrid, which _should_ contribute to reducing fxo echo.
However, the default driver does not currently preload (or dynamically)
adjust the hybrid coefficients to optimize the pstn line interface.
There has been some work going on recently to do that, but I don't
believe the code is production ready as yet. Setting the coefficients
is primarily a near-end echo optimization.

The tdm card currently has another issue that causes at least some
implementations to fail after several days/weeks of operation. There
is some work going on at digium to identify the source of that problem,
however with intermitant problems that only show up over long periods
of time, it is very difficult to catch and identify the root cause.
The failure requires asterisk to be stopped and the card drivers to
be reloaded to correct the issue.

Once the tdm issues have been resolved, its my opinion the card will
be a very cost effective way to interface to pstn lines in the small
office (and home) environment.



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Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

2005-01-30 Thread Tim Mattison

Try the monitor application instead of record.  I think that'll do what
you're looking for.

On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote:
> Hello All,
> 
> I would like to record inbound and outbound calls to and from one
> number.
> 
> I tried to add lines to my extensions.conf:
> 
> DAY=`date "+%m-%d-%y_%H:%m"`
> 
> ;outbound
> exten => 551212,1,Record(${DAY}:gsm)
> exten => 551212,2,Dial(${TRUNKL3}/${EXTEN})
> 
> ;Inbound
> [line2]
> exten => 551212,1,Record(${DAY}:gsm)
> exten => 551212,2,Dial(SIP/101,20)
> exten => 551212,3,Hangup
> 
> 
> 

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Re: [Asterisk-Users] where to buy x100p

2005-01-30 Thread Jon Gabrielson

The digium X100P is probably fine.  What most of these people
are talking about are the X100P clones which are of varying 
qualities.  I have had no problems, but results vary.

Cheers,


Jon.

On Sunday 30 January 2005 09:42 am, dean collins wrote:
> Why does the X100P have echo and the Wildcard TDM400P have no echo?
>
>
>
> I thought the only advantage of using the TDM400P was that it used less
> interrupts than the X100P?
>
> Are there any other advantages?
>
>
>
>
>
>
>
> Cheers,
>
> Dean
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Remco
> Barende
> Sent: Sunday, January 30, 2005 10:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] where to buy x100p
>
> >> Want to buy the X100P clones I have? :)
> >
> > About the best thing that you can do to reduce Echo on the X101P and
>
> X100P
>
> > cards is to;
> >
> >
> >
> > 1. Set the following in zapata.conf;
> >
> >
> >
> > echocancel=128
> >
> > echocancelwhenbridged=no
> >
> > echotraining=800
> >
> >
> >
> > 2. Enable aggressive echo cancellation in zaptel's zconfig.h file;
> >
> >
> >
> > /*
> >
> > * Uncomment for aggressive residual echo supression under
> >
> > * MARK2 echo canceller
> >
> > */
> >
> > #define AGGRESSIVE_SUPPRESSOR
> >
> >
> >
> > I've had much better luck at killing Echo issues with the above
> >
> > suggestions, on both my P133 X101P box as well as my TE405P based PRI
> >
> > Gateways. It doesn't solve all the issues, but it does help quite
> >
> > noticeably.
>
> Thanks for the tip. However, I have been trying evening after avening to
>
>
> get the echo out by tweaking the settings. When I received the TDM11 I
>
> pulled out the X101P clone followed the example config at the Digium
> site
>
> and I was up and running without any. I don't mind trying to play around
>
>
> but wasting a lot of time on really cheap hardware was too much waste of
>
>
> time when I look back :)
>
>
>
> It's probably possible to get the echo out but it's not worth it for me
> to
>
> figure it out :)
>
>
>
>
>
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Re: [Asterisk-Users] Asterisk on MS Virtual Server

2005-01-30 Thread Paul Tyreman

http://www.digium.com/index.php?menu=astwind
I think this may be worth a look, I'm downloading it as I type this 
e-mail...

I didn't know Asterisk had the possibility of being run on a windows machine 
and while it's not as stable as a Linux implementation, it might just do for 
the moment, as I don't have many users.

Is there any documentation on this windows based software, or if not, do you 
know where I can get more info on it ?

Thanks, Paul.

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein
Posted At: 30 January 2005 14:43
Posted To: Asterisk-Users
Conversation: [Asterisk-Users] Asterisk on MS Virtual Server
Subject: Re: [Asterisk-Users] Asterisk on MS Virtual Server

It "works", but you will have timing issues and very poor audio quality.
I've run linux both under Vmware as well as running it under CoLinux 
directly on windows w/ no emulation neccessary. All emulation / 
virtualization layers have a problem whereby they are not able to keep up 
with the interrupt frequency that good quality audio requires, so you will 
hear pops, drops and buzzes in your conversation.

That being said, I did run my Home PBX for several months on a Dual Opteron 
box running Windows XP and the AstWind package:

http://www.digium.com/index.php?menu=astwind
SIP to IAX2 was OK, and only had some minor glitches. However, when you 
start involving disk access, the audio falls apart almost entirely. When I 
asked on the CoLinux mailing list about the reasons for this, they explained 
that they have to "fake" interrupt timings for the CoLinux kernel, and as a 
result they can't gurantee anything consistent for latency between cycles.

--
   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] Re: No ringback over IAX - LiveVoip

2005-01-30 Thread Andrew Kohlsmith

On January 30, 2005 09:31 am, Greg Boehnlein wrote:
> Same reason people stick with Gentoo after a stage one installation. ;) I
> have a theory about Gentoo that explains the rabid nature of Gentoo fans.
> I believe that people that radically defend Gentoo and it's stage one
> installation process are people that have fought through the process and
> gotten a system to work. After spending 2 days working at it, the last
> thing they want to do is admit that they are a total idiot for wasting 48
> hours of their life getting their system to a login prompt, so in a
> classic case of denial, they become raging defenders of the cause. If they
> convince themselves, and others that Gentoo is the best thing since sliced
> bread, they feel better about themselves.

Well if you're doing it for a learning experience that is one thing.  I used 
LFS and scratchbox for those purposes.  :-)

> Now, with crappy VoIP providers it may be that they just do not want to
> let go of the dream. Or, they just want to recover the value of the money
> they have deposited with that company. ;)

Credit cards have a great feature where you can clawback any charge.  Use it 
wisely.  :-)

> P.S. I have no experience with Livevoip or their service, so I have no
> idea if it is crappy or not. However, pretty any much VoIP service
> delivered over the public, non-QOS controlled Internet is going to have
> it's share of problems at some point in time.

Yes and no...  Typically speaking, once you're at your upstream provider's 
router there are no bottlenecks.  It's all in the last mile, in my 
experience.  If your provider's oversubscribing too much then that is another 
issue entirely but typically if you're not on a consumer-grade connection 
there isn't a whole lot of trouble with QoS and the internet, barring the 
next worm.

-A.
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[Asterisk-Users] newlines in application data strings (e.g. userevent)

2005-01-30 Thread Kevin Blackham

exten => s,9,UserEvent(AgentMoreTime,Agent: ${agent}\r\nUntil: ${wrapupat});

Fragment "\r\n" parses into "rn".  "\\r\\n" turns into "\r\n" (uninterpreted).
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