Re: [Asterisk-Users] help voicepulse connect

[Brian West]

Hey dude... they email you the config.. but you might wanna have your
priority numbers correct.

exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},40,tT)
exten => _1NXXNXX,2,Playback,vm-goodbye


On Mon, 17 Nov 2003, Azher Amin wrote:

>
> voicepulse works fine for me ..
>
> In extensions.conf
>
> for usa dialing
> exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},40,tT)
> exten => _1NXXNXX,1,Playback,vm-goodbye
>
> for international dialing
> exten => _1011.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN-1}
> exten => _1011.,1,Playback,vm-goodbye
>
> In iax.conf
>
> [voicepulse]
> context = foo
> secret=secret
> auth=md5
> type=friend
> host=gw5.voicepulse.com
>
> I hope it will work for you as well.
>
> Azher
>
>
> listas iPfone <[EMAIL PROTECTED]> wrote:
> Hi All
>
> I signed up for an account with voicepulse connect service and received the info to 
> set up asterisk.
>
> Anyone have that confs to send as an example?
>
> Thanks
>
> Miklos
>
>
>
> -
> Do you Yahoo!?
> Protect your identity with Yahoo! Mail AddressGuard
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[Asterisk-Users] mysql addon

[Sathya Weerasooriya]

Hello,

I am trying to install the cdr-mysql. Information given in the following
kink is what I am trying to follow;

http://www.voip-info.org/wiki-Asterisk+cdr+mysql

I cant figure out where to install the asterisk-addons. Is it in /usr/src or
/usr/src/asterisk ?

Once I create the cdr-mysql.conf. Is there a command to initiate the mysql
module, or just a reload of a config is enough ?

Thanks a bunch

Sathya


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[Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Andrew Nelson]

Maybe someone here has found a good solution to this problem.

I voulenteer with a local Search And Rescue unit and I was speaking with the 
senior members about how they interface the command trailer PBX with the PSTN 
or cellular networks when they are on scene at a remote location.  Turns out 
they don't.  Thus that got me to thinking about how one would get Asterisk to 
interface with a cell phone directly, or what hardware out there works well 
for the task.  I have managed to figure out how to make my Motorola StarTAC 
make outgoing voice calls for instance and making an interface for this phone 
should be quite possible, but it would be nice to have incoming phone calls 
too.  I know there are many companies out there which make such producs, and 
that a Google search can yield many results, I want thoughts and opinions, 
stuff which Google suks at.

Thanks!

-Andrew
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Re: [Asterisk-Users] mysql addon

[WipeOut]

Sathya Weerasooriya wrote:

Hello,

I am trying to install the cdr-mysql. Information given in the following
kink is what I am trying to follow;
http://www.voip-info.org/wiki-Asterisk+cdr+mysql

I cant figure out where to install the asterisk-addons. Is it in /usr/src or
/usr/src/asterisk ?
Once I create the cdr-mysql.conf. Is there a command to initiate the mysql
module, or just a reload of a config is enough ?
Thanks a bunch

Sathya

 

Make sure you have mysql and mysql-devel packages installed..

then..

# cd /usr/src
# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
# cvs login - the password is anoncvs.
# cvs checkout asterisk-addons
# cd asterisk-addons
# make install
That should be it..

Later..





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[Asterisk-Users] SIP silence detection

[Rattana BIV]

Hi;
 
 
Just a little question about SIP.
 
Is there silence detection with SIP ?
If yes can I suppress it ?
I use asterisk with SJPhone and I think there 
silence detection or maybe my ear doesn't hear well :)
 
 
 
 
Regards
 
 
Rattana

RE: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Vledder, Hans]

Hi Andrew,

I have a similar challenge. I will have to connect a remote location with
PBX to a central location with PBX. While roaming the Internet I came
accross this:

http://www.nokia.com/nokia/0,8764,43170,00.html

Two PSTN/GSM gateways called the Nokia 22 and the Nokia 32. I don't know how
it will operate with * yet but it seems rather transparant.

Regards,
Hans

-Original Message-
From: Andrew Nelson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 9:35 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network


Maybe someone here has found a good solution to this problem.

I voulenteer with a local Search And Rescue unit and I was speaking with the

senior members about how they interface the command trailer PBX with the
PSTN 
or cellular networks when they are on scene at a remote location.  Turns out

they don't.  Thus that got me to thinking about how one would get Asterisk
to 
interface with a cell phone directly, or what hardware out there works well 
for the task.  I have managed to figure out how to make my Motorola StarTAC 
make outgoing voice calls for instance and making an interface for this
phone 
should be quite possible, but it would be nice to have incoming phone calls 
too.  I know there are many companies out there which make such producs, and

that a Google search can yield many results, I want thoughts and opinions, 
stuff which Google suks at.

Thanks!

-Andrew
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Re: [Asterisk-Users] MP3Player problem -repost

[Areski]

Hi Ryan,

Thanks for your help, that works now ;P
Aresk


On Mon, 2003-11-17 at 23:52, Ryan Tucker wrote:
> On 17 Nov 2003 13:39:20 +0100, Areski <[EMAIL PROTECTED]> wrote:
> > I tried also to enter directly this instruction:
> > mpg123 -w ki.wav http://digitaljukebox.com/Carta.mp3
> > And I get :
> > HTTP request failed: HTML PUBLIC "-//IETF//DTD HTML 2.0//EN">
> >
> > The file exist, I get do a wget on it...
> > Some ideas how to get it working ???
> 
> It looks like it's doing a redirect from digitaljukebox.com to 
> www.digitaljukebox.com:
> 
> [EMAIL PROTECTED]:~$ telnet digitaljukebox.com 80
> Trying 216.98.141.3...
> Connected to 4h1413.aspadmin.net.
> Escape character is '^]'.
> HEAD /Carta.mp3 HTTP/1.0
> Host: digitaljukebox.com
> 
> HTTP/1.1 302 Found
> Date: Mon, 17 Nov 2003 22:48:25 GMT
> Server: Apache/1.3.20 Sun Cobalt (Unix) mod_ssl/2.8.4 OpenSSL/0.9.6b 
> PHP/4.0.6 mod_auth_pam_external/0.1 FrontPage/4.0.4.3 mod_perl/1.25
> Location: http://www.digitaljukebox.com/Carta.mp3
> Connection: close
> Content-Type: text/html; charset=iso-8859-1
> 
> Connection closed by foreign host.
> 
> Try using www.digitaljukebox.com instead.  mpg123 has a simple mind and 
> likes things to be simple HTTP-wise.  -rt

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[Asterisk-Users] double-dial in SIP Grandstream

[Marc SCHAEFER]

Hi,

I have even now connected to IAXtel at number 1-700-895-5211
when I am in the office, so Asterisk is great.

I just found something strange, which is that if I am already in a
connection with my Grandstream and talking, and a second call comes in,
it rings on the Grandstream.

However, if I am not talking but waiting for dialing, the caller gets a
busy signal (good).

How can I make sure there is only one call at a time to the SIP phone ?
(call waiting could be useful, but I didn't figure out how to do this
with the SIP).


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Re: [Asterisk-Users] VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk

[Rich Adamson]

> > Surprise, people.  While geeks may be all in favor of used equipment
> > (and yes, most of it is probably no worse than new equipment), there are
> > many customers who are uncomfortable with buying used equipment,
> > probably because many of them have gotten burned in the past.

I probably shouldn't be contributing to this almost worthless thread,
however there are "some" accounting types that do not want to deal with
used equipment regardless of the source; there are some US accounting
rules/guidelines that address it. But I'd have to guess that most on 
this list aren't aware of those things, and likely for the same reason
why geeks aren't involved with the business decision as to which pbx 
their company considers for replacements, etc.


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[Asterisk-Users] Re: DMTF tones when VOIP call comes in

[jaycard]

I keep getting a error message every time a call comes in via a VOIP 
source ..

" NOTICE[262161]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 
19 received "

Every time I enter a digit asterisk produces on line of this 
error/notice, but it is recognizing the numbers correctly though. 
Through my X100p card it doesnt produce this notice.
It doesnt matter which voip account, IAX,Iconnect,FWD. Same message... 
any ideas?

Jaycard.

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Re: [Asterisk-Users] mysql addon

[jaycard]

WipeOut wrote:

Sathya Weerasooriya wrote:

Hello,

I am trying to install the cdr-mysql. Information given in the following
kink is what I am trying to follow;
http://www.voip-info.org/wiki-Asterisk+cdr+mysql

I cant figure out where to install the asterisk-addons. Is it in 
/usr/src or
/usr/src/asterisk ?

Once I create the cdr-mysql.conf. Is there a command to initiate the 
mysql
module, or just a reload of a config is enough ?

Thanks a bunch

Sathya

 

Make sure you have mysql and mysql-devel packages installed..

then..

# cd /usr/src
# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
# cvs login- the password is anoncvs.
# cvs checkout asterisk-addons
# cd asterisk-addons
# make install
That should be it..

Later..





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You might also want to install the agi perl script packages if you want 
to work with
link asterisk with perl - mysql. Heres the website:

http://asterisk.gnuinter.net/

Jaycard.

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Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Rich Adamson]

> I voulenteer with a local Search And Rescue unit and I was speaking with the 
> senior members about how they interface the command trailer PBX with the PSTN 
> or cellular networks when they are on scene at a remote location.  Turns out 
> they don't.  Thus that got me to thinking about how one would get Asterisk to 
> interface with a cell phone directly, or what hardware out there works well 
> for the task.  I have managed to figure out how to make my Motorola StarTAC 
> make outgoing voice calls for instance and making an interface for this phone 
> should be quite possible, but it would be nice to have incoming phone calls 
> too.  I know there are many companies out there which make such producs, and 
> that a Google search can yield many results, I want thoughts and opinions, 
> stuff which Google suks at.

Some of the older cell phones use to expose either a 2-wire or 4-wire
interface via a connector on the phone (don't know about transmission
levels), but not sure what the more current models are doing. Nokia and 
Motorola sponsor application development lists, free development tools, 
limited access to knowledgable engineers, compatibility test centers,
and other such technical references that are open for anyone to join.
I'd start by digging around Motorola's web site (they do have references
to the lists if dig deep enough).



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[Asterisk-Users] Asterisk Festival Perl Net::POP3

[Bartosz Jozwiak]

Hello,
 
Did somebody tried to make a script to check e-mail 
with POP3?
And pass it to Asterisk Festval ?
 
Bart.

Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Linus Surguy]

> > I voulenteer with a local Search And Rescue unit and I was speaking with
the
> > senior members about how they interface the command trailer PBX with the
PSTN
> > or cellular networks when they are on scene at a remote location.  Turns
out
> > they don't.  Thus that got me to thinking about how one would get
Asterisk to
> > interface with a cell phone directly, or what hardware out there works
well
> > for the task.  I have managed to figure out how to make my Motorola
StarTAC
> > make outgoing voice calls for instance and making an interface for this
phone
> > should be quite possible, but it would be nice to have incoming phone
calls
> > too.  I know there are many companies out there which make such producs,
and
> > that a Google search can yield many results, I want thoughts and
opinions,
> > stuff which Google suks at.
>
> Some of the older cell phones use to expose either a 2-wire or 4-wire
> interface via a connector on the phone (don't know about transmission

I won't bother with any of that - purchase a Nokia Premicell (or other
manufacturers similar item). This device takes a normal GSM SIM card and
then presents a normal PSTN line interface - plug that into your normal
Asterisk PSTN line card - job done.

Linus


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Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Andrew Thompson]

I wasn't able to figure out why they have two models, but basically, you take your SIM 
card out of your GSM phone and put it in the device. It turns your cellphone into a 
FXO(?). You'll need a X100p or a channel bank to plug it into your asterisk.

Sucks if you use CDMA...

-
Andrew Thompson

- Original Message - 
From: "Vledder, Hans" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 18, 2003 4:24 AM
Subject: RE: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network


> Hi Andrew,
> 
> I have a similar challenge. I will have to connect a remote location with
> PBX to a central location with PBX. While roaming the Internet I came
> accross this:
> 
> http://www.nokia.com/nokia/0,8764,43170,00.html
> 
> Two PSTN/GSM gateways called the Nokia 22 and the Nokia 32. I don't know how
> it will operate with * yet but it seems rather transparant.
> 
> Regards,
> Hans
> 
> -Original Message-
> From: Andrew Nelson [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, November 18, 2003 9:35 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network
> 
> 
> Maybe someone here has found a good solution to this problem.
> 
> I voulenteer with a local Search And Rescue unit and I was speaking with the
> 
> senior members about how they interface the command trailer PBX with the
> PSTN 
> or cellular networks when they are on scene at a remote location.  Turns out
> 
> they don't.  Thus that got me to thinking about how one would get Asterisk
> to 
> interface with a cell phone directly, or what hardware out there works well 
> for the task.  I have managed to figure out how to make my Motorola StarTAC 
> make outgoing voice calls for instance and making an interface for this
> phone 
> should be quite possible, but it would be nice to have incoming phone calls 
> too.  I know there are many companies out there which make such producs, and
> 
> that a Google search can yield many results, I want thoughts and opinions, 
> stuff which Google suks at.
> 
> Thanks!
> 
> -Andrew
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> 
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> contains information that may be confidential. Unless you are the named
> addressee or an authorized designee, you may not copy or use it, or disclose
> it to anyone else. If you received it in error please notify us immediately
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Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Dan]

Hi,

...
> I won't bother with any of that - purchase a Nokia Premicell (or other
> manufacturers similar item). This device takes a normal GSM SIM card and
> then presents a normal PSTN line interface - plug that into your normal
> Asterisk PSTN line card - job done.

A PCI and/or USB device, Asterisk compatible, able to accept a SIM card and
talking digital only (inclusiv audio) will be a great thing, even internally
it can be based on a cheaper GSM phone...

What about a PCMCIA GSM card connected through a PCI/PCMCIA adapter?
There is only needed an Asterisk supported driver.

:-)
Dan

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[Asterisk-Users] Bayonne and Asterisk

[Dirk-Jan Wemmers]

All,

is anyone using Bayonne in conjunction with Asterisk? I'm currently using 
only Bayonne, but I'm investigating the possibilities of switching the 
telephony frontend over to Asterisk, and have Asterisk route the IVR tasks 
to Bayonne through H323.

Anyone care to share his views on this approach? Any pointers or do's  and 
don'ts? All info is greatly appreciated!

Regards,
Dirk-Jan
--
Dirk-Jan Wemmers, Capcave B.V.
Zonnebaan 17, 3542EA Utrecht
T +31(0)30-2149670, F +31(0)30-2149679
M +31(0)651 063040, E [EMAIL PROTECTED]
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[Asterisk-Users] Notice with asterisk System application

[Rattana BIV]

Hi,
 
I notice something with asterisk with the System 
application.
When I lauch asterisk with -c option the 
application System work correctly.
But when I lauch asterisk without option, the 
application System doesn't lauch  command.
It is normal ?
 
 
Regards
Rattana

Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[David Uzzell]

Dan wrote:
Hi,

...

I won't bother with any of that - purchase a Nokia Premicell (or other
manufacturers similar item). This device takes a normal GSM SIM card and
then presents a normal PSTN line interface - plug that into your normal
Asterisk PSTN line card - job done.


A PCI and/or USB device, Asterisk compatible, able to accept a SIM card and
talking digital only (inclusiv audio) will be a great thing, even internally
it can be based on a cheaper GSM phone...
What about a PCMCIA GSM card connected through a PCI/PCMCIA adapter?
There is only needed an Asterisk supported driver.
:-)
Dan
That sounds like a really GREAT idea, I thought I would put my foot in.

A great result in this would be some way to drive the Nokia phones 
through the serial Port and Computer cable!

This would allow us in Australia to be able to use CDMA it would also 
mean that any cheap Nokia phone could be used as long as it has the 
computer cable link.

Anyway thats my 2c worth in the Pot!

David



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RE: [Asterisk-Users] double-dial in SIP Grandstream

[Bisker, Scott (7805)]

Marc,

This is the typical behavior for call waiting.  While you are initiating a
call, people who call your number will get a busy signal until your first
call connects.  Once the call connects, the number 2 caller will hear a ring
until you pickup.  

If you want to disable callwaiting then put "callwaiting=no" in sip.conf for
that particular alias.

[]
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[Asterisk-Users] App Queue

[Josh Edwards]

Does anyone have a good HOWTO on queues Is your computer infected with a virus?  Find out with a FREE computer virus scan from McAfee.  Take the FreeScan now! 
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[Asterisk-Users] Low Volume X100P

[Kevin]

Has anyone experienced low volume with
the X100P FXO card?

Re: [Asterisk-Users] Low Volume X100P

[Josh J. Zuerner]

Yes, but, just increase the rxgain in your 
zapata.conf and it will likely take care of the issue.
 
Josh

  - Original Message - 
  From: 
  Kevin 
  To: [EMAIL PROTECTED] 
  
  Sent: Tuesday, November 18, 2003 9:12 
  AM
  Subject: [Asterisk-Users] Low Volume 
  X100P
  
  
  
  Has anyone 
  experienced low volume with the X100P FXO 
  card?

Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0

[Andrew Thompson]

Tried two different Win2k systems and it crashes on load.

-
Andrew Thompson
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is 
to watch the cursor blink. Close your eyes. The opinions stated above are yours. You 
cannot imagine why you ever felt otherwise.

- Original Message - 
From: "Michael Van Donselaar" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 17, 2003 6:45 PM
Subject: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0


iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux.  Win32
and Linux binaries as well as the LGPL source are available at:

http://iaxclient.sourceforge.net

Recent improvements are a less cluttered user interface, audible ringback and
audible outgoing ring, and of course IAX2 protocol support.

iaxComm is based upon the wxWindow GUI framework and compiles on Microsoft
Windows, Linux, and OS X.

It runs on Windows XP, Windows 2000, and Red Hat 9.0.  It doesn't (yet) run on
OS X.

The digium S100U is supported for handset audio only (no hook state detection,
no DTMF decoding, no ringing) with the IPO-11 audio drivers.  I've been trying
to get a Windows SDK, but no luck so far.

Please send me bug reports, critiques, and feature requests.  I would like to
get a 0.99 release out by 1 DEC 03.

If anyone is interested, there is a mailing list for the iaxclient library upon
which iaxComm is written.

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RE: [Asterisk-Users] Bayonne and Asterisk

[mattf]

I used Bayonne for 2 years before switching to Asterisk. Right now I'm still
running Bayonne on one application and it's been running happily without me
looking at it for over 6 months. I'd say these are 

the strengths of Bayonne:
- Runs on Dialogic, Pika and other widely available hardware
- extremely reliable, mine never crashes

and here are the weaknesses:
- nowhere near as active of a support community as Asterisk has
- configuration of the hardware/drivers is a nightmare compared to
Asterisk/Digium
- it is quite limited in it's included apps, IVR and voicemail
- not as many options for scripting as Asterisk
- it was not designed to have full PBX functionality, some PBX functionality
is added as afterthought
- the code/organization/flow is not as well thought out or documented as
Asterisk is

And yes, they can run fine together(I'm not using VOIP, just a T1 out of
Asterisk to Bayonne to test and see if it would work). The IVR application
that I currently still have running on Bayonne is only still on Bayonne
because it can never go down, and Bayonne has proven itself to me to be
extremely stable, while I cannot personally say AT THIS TIME that an
Asterisk box would stay up for over 6 months with no crashes.

At last check I was never able to get VOIP inbound working on Bayonne, maybe
this has changed in the last 6 months but if you do get it working I'd be
interested to find out how.


MATT---



-Original Message-
From: Dirk-Jan Wemmers [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 8:44 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Bayonne and Asterisk


All,

is anyone using Bayonne in conjunction with Asterisk? I'm currently using 
only Bayonne, but I'm investigating the possibilities of switching the 
telephony frontend over to Asterisk, and have Asterisk route the IVR tasks 
to Bayonne through H323.

Anyone care to share his views on this approach? Any pointers or do's  and 
don'ts? All info is greatly appreciated!

Regards,
Dirk-Jan

-- 
Dirk-Jan Wemmers, Capcave B.V.

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[Asterisk-Users] ask problem about softphone--asterisk--softphone, Urgent!!!

[Qian Lv]

Hi, all,
I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below:
 
Softphone1<-->Asterisk SIP<>Softphone2
(User Agent)       (Proxy)        (User Agent)
155.69.xx.xx 155.69.yy.yy     155.69.zz.zz    
  zhou   mysipproxy.com  Reltec
 
If I use softphone1(zhou) to dial softphone2(Reltec) directly, not accroess Asterisk SIP (proxy), it can work. But when I use asterisk SIP as a proxy, then Softphone1(zhou) shows "Not Found", and it seems it can not find softphone2's address.
 
It seems an easy problem, but it waste me about one week's time. The main content in my [sip.conf] file is:
  ...  
   [general]
 port=5060
 bindaddr=0.0.0.0
 context=bogon-calls
 allow=all
    
  [mysipproxy.com]
 type=friend
 host=155.69.yy.yy
 fromuser=lq
 
 [zhou]
 type=friend
 host=dynamic
 defaultip=155.69.xx.xx
 context=from-sip
 fromdomain=mysipproxy.com
 
 [Raytec]
 type=friend
 host=dynamic
 defaultip=155.69.zz.zz
 context=from-sip
 fromdomain=mysipproxy.com
 
The main content in my [extensions.conf] is:
  ...
 [bogon-calls]
    exten=>_.,1,Congestion
 [from-sip]
    exten => 1, 1, Dial(SIP/zhou,20)
    exten => 1, 102, Hangup

    exten => 2, 1, Dial(SIP/Raytec,20)
    exten => 2, 102, Hangup 
The result in asterisk SIP is listed below:
 
*CLI> sip debugSIP Debugging Enabled*CLI> Sip read:INVITE sip:[EMAIL PROTECTED] SIP/2.0Call-ID: [EMAIL PROTECTED]Content-Length: 125Content-Type: application/sdpTo: sip:[EMAIL PROTECTED]From: sip:[EMAIL PROTECTED];tag=74763707Contact: sip:[EMAIL PROTECTED]:5060CSeq: 1 INVITEVia: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C400F8EF064B14-33*0
v=0o=- 1069165096343 1069165096343 IN IP4 155.69.149.113s=-c=IN IP4 155.69.149.113t=0 0m=audio 5006 RTP/AVP 3 0 8
9 headers, 6 linesUsing latest request as basis requestSending to 155.69.149.113 : 5060 (non-NAT)Found audio format UNKNFound audio format UNKNFound audio format ALAWCapabilities: us - 2147483647, them - 14/0, combined - 14Non-codec capabilities: us - 1, them - 0, combined - 0Looking for Raytec in from-sipTransmitting (no NAT):SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C400F8EF064B14-33*0From: sip:[EMAIL PROTECTED];tag=74763707To: sip:[EMAIL PROTECTED];tag=as1d9111e1Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Content-Length: 0
 to 155.69.149.113:5060Sip read:ACK sip:[EMAIL PROTECTED] SIP/2.0From: sip:[EMAIL PROTECTED];tag=74763707To: sip:[EMAIL PROTECTED];tag=as1d9111e1Call-ID: [EMAIL PROTECTED]CSeq: 1 ACKVia: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C400F8EF064B14-33*0Content-Length: 0
7 headers, 0 lines
 
 

 
 
 
 
 
 
 
 
 
 
It seems the extensions.conf has some problem, but I don't know how to write the correct dialplan. Any suggestions will be appreciated.
Thanks.
 
 
 Regards,===Lv Qian,Ph.D Student,School of Computer Engineering,Nanyang Technological University,Singapore 639798=
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[Asterisk-Users] ask problem about softphone--asterisk--softphone, Urgent!!!

[Qian Lv]

Hi, all,
I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below:
 
Softphone1<-->Asterisk SIP<>Softphone2
(User Agent)       (Proxy)        (User Agent)
155.69.xx.xx 155.69.yy.yy     155.69.zz.zz    
  zhou   mysipproxy.com  Reltec
 
If I use softphone1(zhou) to dial softphone2(Reltec) directly, not accroess Asterisk SIP (proxy), it can work. But when I use asterisk SIP as a proxy, then Softphone1(zhou) shows "Not Found", and it seems it can not find softphone2's address.
 
It seems an easy problem, but it waste me about one week's time. The main content in my [sip.conf] file is:
  ...  
   [general]
 port=5060
 bindaddr=0.0.0.0
 context=bogon-calls
 allow=all
    
  [mysipproxy.com]
 type=friend
 host=155.69.yy.yy
 fromuser=lq
 
 [zhou]
 type=friend
 host=dynamic
 defaultip=155.69.xx.xx
 context=from-sip
 fromdomain=mysipproxy.com
 
 [Raytec]
 type=friend
 host=dynamic
 defaultip=155.69.zz.zz
 context=from-sip
 fromdomain=mysipproxy.com
 
The main content in my [extensions.conf] is:
  ...
 [bogon-calls]
    exten=>_.,1,Congestion
 [from-sip]
    exten => 1, 1, Dial(SIP/zhou,20)
    exten => 1, 102, Hangup

    exten => 2, 1, Dial(SIP/Raytec,20)
    exten => 2, 102, Hangup 
The result in asterisk SIP is listed below:
 
*CLI> sip debugSIP Debugging Enabled*CLI> Sip read:INVITE sip:[EMAIL PROTECTED] SIP/2.0Call-ID: [EMAIL PROTECTED]Content-Length: 125Content-Type: application/sdpTo: sip:[EMAIL PROTECTED]From: sip:[EMAIL PROTECTED];tag=74763707Contact: sip:[EMAIL PROTECTED]:5060CSeq: 1 INVITEVia: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C400F8EF064B14-33*0
v=0o=- 1069165096343 1069165096343 IN IP4 155.69.149.113s=-c=IN IP4 155.69.149.113t=0 0m=audio 5006 RTP/AVP 3 0 8
9 headers, 6 linesUsing latest request as basis requestSending to 155.69.149.113 : 5060 (non-NAT)Found audio format UNKNFound audio format UNKNFound audio format ALAWCapabilities: us - 2147483647, them - 14/0, combined - 14Non-codec capabilities: us - 1, them - 0, combined - 0Looking for Raytec in from-sipTransmitting (no NAT):SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C400F8EF064B14-33*0From: sip:[EMAIL PROTECTED];tag=74763707To: sip:[EMAIL PROTECTED];tag=as1d9111e1Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Content-Length: 0
 to 155.69.149.113:5060Sip read:ACK sip:[EMAIL PROTECTED] SIP/2.0From: sip:[EMAIL PROTECTED];tag=74763707To: sip:[EMAIL PROTECTED];tag=as1d9111e1Call-ID: [EMAIL PROTECTED]CSeq: 1 ACKVia: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C400F8EF064B14-33*0Content-Length: 0
7 headers, 0 lines
 
 

 
 
 
 
 
 
 
 
 
 
It seems the extensions.conf has some problem, but I don't know how to write the correct dialplan. Any suggestions will be appreciated.
Thanks.
 
 
 Regards,===Lv Qian,Ph.D Student,School of Computer Engineering,Nanyang Technological University,Singapore 639798=
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Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Dan]

Hi,

- Original Message - 
From: "David Uzzell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 18, 2003 3:54 PM
Subject: Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network


> Dan wrote:
> > Hi,
> >
> > ...
> >
> >>I won't bother with any of that - purchase a Nokia Premicell (or other
> >>manufacturers similar item). This device takes a normal GSM SIM card and
> >>then presents a normal PSTN line interface - plug that into your normal
> >>Asterisk PSTN line card - job done.
> >
> >
> > A PCI and/or USB device, Asterisk compatible, able to accept a SIM card
and
> > talking digital only (inclusiv audio) will be a great thing, even
internally
> > it can be based on a cheaper GSM phone...
> >
> > What about a PCMCIA GSM card connected through a PCI/PCMCIA adapter?
> > There is only needed an Asterisk supported driver.
> >
> > :-)
> > Dan
> >
>
> That sounds like a really GREAT idea, I thought I would put my foot in.
>
> A great result in this would be some way to drive the Nokia phones
> through the serial Port and Computer cable!

...then you need an audio cable too, connected to the handsfree socket

What about then to use the OSS driver and a cheaper full duplex sound card
to connect the phone
You neet then just the following:
- a GSM/CDMA phone with handsfree socket and data cable
- a full duplex sound card
- OSS driver for the sound part (* CONSOLE maybe???)
- a small driver for the serial port

>
> This would allow us in Australia to be able to use CDMA it would also
> mean that any cheap Nokia phone could be used as long as it has the
> computer cable link.
>

BR,
Dan

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[Asterisk-Users] DIGI Datafire QuadMicro

[Michael Devenijn]

Did anybody tried this card with asterisk ?
 
http://www.digi.com/pdf/prd_mca_datafirequad.pdf
 
 

Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0

[Dan]

Hi,

- Original Message - 
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 18, 2003 4:36 PM
Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP,
Red Hat 9.0


> Tried two different Win2k systems and it crashes on load.
>


Tried on WinXP Pro and it loads, but in the background (no window).
There is something needed from the wxWindows package to just run the
executable?

BR,
Dan

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[Asterisk-Users] Hard & soft phones

[Michael Graves]

Hello All,

I'm about 1/2 way through building an * system for home office. I have
the server built, * installed, but I'm waiting for a Grandstream phone
to arrive to have a reliable client. In the mean time I have a few
short questions.

1. I have the X-Pro soft phone client. It's presently configured for
FWD. What settings would I need in it to get it to use the * server? I
have * logging into FWD already. FWIW, the * server is 192.168.1.30.

2. Zultys 4x4 or SNOM 200 - anyone here have experience with these hard
phones? I'm pondering which hard phones to use. I need 2 full featured
office phones and 2 or 3 simple phones.

3. FXS & Analog for cordless phone or Wifi SIP phone? What are the
advantages/disadvantages?

Thanks,

Michael Graves
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc.  [EMAIL PROTECTED]
 FWD 54245

"Kick at the darkness 'till it bleeds daylight" - Bruce Cockburn
 
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Re: [Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode

[Daniel ANDRE]

Hello,
I have set one function key of my IP10s to send the flash event and it
seems to work but only for blind transfert and not for consultative
transfert or call conferencing. How should I do to use the flash key to
do that?

Regards,

Daniel 

Florian Overkamp wrote:

  Hi,
 

  
  
-Original Message-
I have added these lines and still no transfert menu on my IP10S

  
  
Can you try what happens if you program one of the function keys to send
'Flash' and use that ?

I have not tried this, but I can imagine it might work that way..

Florian

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-- 
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Serveur kwartz - http://www.kwartz.com

RE: [Asterisk-Users] Global configuration question

[Sérgio Bernardo]

Hi Roy, 

Thanks a lot for your answer. You gave-me a completely new direction on
the available solutions... Why didn't I found those IP phones before???
Now it starts being interesting ;-)

--
Sérgio

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Roy Sigurd Karlsbakk
> Sent: quinta-feira, 13 de Novembro de 2003 20:12
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Global configuration question
> 
> 
> > What hardware should I use for the telephones ? IP Phones seam too
> > expensive and I'm sure they do a lot of things that are not 
> needed in
> > Asterisk context...
> 
> We've been using dlink dph-100, snom 100 and grandstream 
> budgetone 100. 
> We've ditched the dlink phones, as they're too unreliable (probably 
> hardware problem - tried several software versions). We're using snom 
> 100 phones for the management and sales staff, and budgettone (or 
> barbietone, as someone on irc just called it :) for the tech people. 
> this works well, and as the grandstream phones sell for $65 a 
> piece, I 
> can't really see the big point of not going this way. If you want to 
> use analog phones, you'll need a channel bank (or two) and an extra 
> T100P card (or two - or a TE410P). This is probably not the cheapest 
> way to go - I'd recommend the grandstream phones. We're using 
> it all on 
> the same network as video streaming (both unicast and multicast) with 
> el cheapo managed dlink switches, and it works flawlessly :)
> 
> roy
> 
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> 

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Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0

[Michael Van Donselaar]

On Tue, 18 Nov 2003 09:36:23 -0500, you wrote:

>Tried two different Win2k systems and it crashes on load.

You currently have to run it from the directory it is located in.  If you make a
shortcut, make sure that "Run In:" has the correct directory.

iaxComm needs to see the resources in ${cwd}/rc

I hope that fixes it.  Please let me know
>
>-
>Andrew Thompson
>Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it 
>is to watch the cursor blink. Close your eyes. The opinions stated above are yours. 
>You cannot imagine why you ever felt otherwise.
>
>- Original Message - 
>From: "Michael Van Donselaar" <[EMAIL PROTECTED]>
>To: <[EMAIL PROTECTED]>
>Sent: Monday, November 17, 2003 6:45 PM
>Subject: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
>
>
>iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux.  Win32
>and Linux binaries as well as the LGPL source are available at:
>
>http://iaxclient.sourceforge.net
>
>Recent improvements are a less cluttered user interface, audible ringback and
>audible outgoing ring, and of course IAX2 protocol support.
>
>iaxComm is based upon the wxWindow GUI framework and compiles on Microsoft
>Windows, Linux, and OS X.
>
>It runs on Windows XP, Windows 2000, and Red Hat 9.0.  It doesn't (yet) run on
>OS X.
>
>The digium S100U is supported for handset audio only (no hook state detection,
>no DTMF decoding, no ringing) with the IPO-11 audio drivers.  I've been trying
>to get a Windows SDK, but no luck so far.
>
>Please send me bug reports, critiques, and feature requests.  I would like to
>get a 0.99 release out by 1 DEC 03.
>
>If anyone is interested, there is a mailing list for the iaxclient library upon
>which iaxComm is written.
>
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[Asterisk-Users] Broken pipe

[Peter Zeltins]

About once every day my * goes nuts and "asterisk 
-r" responds with "broken pipe". All calls are dropped immediately, even 
extension 600 (echo). Killing the process and restarting asterisk helps... until 
next day. I'm running 0.5.0 release on RH9. Any ideas what's wrong, and what can 
I do to troubleshoot it?
 
TIA,
Peter

Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Jorge Mendoza]

We are using a CDMA/PSTN adapter (Motorola 800SC) interconnecting the 
cellular network to our pabx. In our case the application is different: 
the cell-to-cell calls are cheaper that pstn <=> cell calls. A fxo 
interface in required your pabx.

Jorge

Andrew Thompson wrote:

I wasn't able to figure out why they have two models, but basically, you take your SIM card out of your GSM phone and put it in the device. It turns your cellphone into a FXO(?). You'll need a X100p or a channel bank to plug it into your asterisk.

Sucks if you use CDMA...

-
Andrew Thompson
- Original Message - 
From: "Vledder, Hans" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 18, 2003 4:24 AM
Subject: RE: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

 

Hi Andrew,

I have a similar challenge. I will have to connect a remote location with
PBX to a central location with PBX. While roaming the Internet I came
accross this:
http://www.nokia.com/nokia/0,8764,43170,00.html

Two PSTN/GSM gateways called the Nokia 22 and the Nokia 32. I don't know how
it will operate with * yet but it seems rather transparant.
Regards,
Hans
-Original Message-
From: Andrew Nelson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 9:35 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network
Maybe someone here has found a good solution to this problem.

I voulenteer with a local Search And Rescue unit and I was speaking with the

senior members about how they interface the command trailer PBX with the
PSTN 
or cellular networks when they are on scene at a remote location.  Turns out

they don't.  Thus that got me to thinking about how one would get Asterisk
to 
interface with a cell phone directly, or what hardware out there works well 
for the task.  I have managed to figure out how to make my Motorola StarTAC 
make outgoing voice calls for instance and making an interface for this
phone 
should be quite possible, but it would be nice to have incoming phone calls 
too.  I know there are many companies out there which make such producs, and

that a Google search can yield many results, I want thoughts and opinions, 
stuff which Google suks at.

Thanks!

-Andrew
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[Asterisk-Users] codec problems between * and cisco hardware h323

[Thomas Haeger]

Hi all,

we use * as a voip gateway using h323 (chan_h323).
We purchased the g729 codec from digium for 10 channels.

But, if comes in a call from a cisco gatway or something similar, there're
problems with the codecs.
If such a call comes in following error occures :

Pitch 1st subfr. !! Wrong Pitch 1st subfr. !! Wrong Pitch 1st subfr.
!! Wrong Pitch 1st subfr. !! Wrong Pitch 1st subfr. !! Wrong
Pitch 1st subfr. !! Wrong Pitch 1st subfr. !! Wrong Pitch 1st subfr.
!! Wrong Pitch 1st subfr. !! Wrong Pitch 1st subfr. !! Wrong
Pitch 1st subfr. !! Wrong Pitch 1st subfr. !! Wrong Pitch 1st subfr.
!! Wrong Pitch 1st subfr. !! Wrong Pitch 1st subfr. !! Wrong
Pitch 1st subfr. !! Wrong Pitch 1st subfr. !! Wrong Pitch 1st subfr.
!! Wrong Pitch 1st subfr. !! Wrong Pitch 1st subfr. !! Wrong
Pitch 1st subfr. !! Wrong Pitch 1st s-- Hungup 'Zap/1-1'h 1st subfr.

I think these messages are from the codec_g729 module...

Knows anybody how can i get both, cisco and *, works together ?

Can somebody help ?

Thanks,
Thomas.

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[Asterisk-Users] Swissvoice ip10s MGCP questions and experiences

[Philipp von Klitzing]

Hi there,

here some questions and experiences after playing for one day with 3 
Swissvoice ip10s and the latest * CVS:

QUESTIONS:
- what is the user option "enter voice mail number" good for? It doesn't 
appear to be of any practical use
- does anyone have some Swissvoice info that I cannot find on their web 
site like the guide to MGCP XML (.svd), guide to configuration file 
format (.cfg), guide to phone configuration through TELNET?
- do I need two lines ("aaln/1" and "aaln/2") or just one in my mgcp.conf 
(since the ip10 appears to support two lines)?
- are there any specs available for the format of the operator/ user 
logo? Must this be b/w, or can it be greyscale (2 bit/ 4 bit/ 8 bit?). As 
I understand both .gif and .jpg are fine. Pixel size?
- how can I obtain newer firmware that hopefully solves my crash problem?

WORKING:
- simple calls, phone book, hands-free
- MWI: F4 key lights up when messages are available
- pick-up: Assign the pick-up function together with *6 to F2.
- assign FLASHING to F1
- DoNotDisturb (on F3) works as expected
- blind transfer using #

NOT or PARTIALLY WORKING:
- I can only set the "phone name" via display --> free idle text?
- with F4 configured as voicemail function: pressing that key dials "#" 
and I must modify extension.conf to point # to VoiceMailMain2? Is there 
no way to change the number dialed by the voicemail button?
- MWI: the button lights up/extinguishes only when some kind of phone 
activity occurs; without any activity it might actually indicate wrong 
info (e.g. when user already check vm via web)
- during startup the phone nicely display the "waiting for call agent" 
message; however when the call agent (call manager?) * disappears 
afterwards the phones do not display that waiting message... :-(
- is there any way to make the "service key" do anything useful? 
Currently it has no function at all.
- crash: let the two ip10 ring the third ip10 at the same time (without 
lifting the handset) - in the moment the 2nd phone is calling the third 
phone crashes and must be rebooted. I can reproduce this 100% with any of 
the three phones. :-(
- the three call forward (CF) functions that can be assigned to the 
function keys F1-F4 don't appear to be useful. Instead of arrange an 
internal phone call forward this will immediately (!) dial the forward 
number entered at the very moment this function is activiated.
- I can successfully use FLASHING (F1) for consultative transfer, however 
I get into bad trouble if the consulted extension does not like to take 
the call and hangs up --> there is apparently no way to get back to my 
original caller, although I see both lines in the display and can move my 
cursor up and down on them. This might also be an * problem.
- web interface: "save current phone settings as profile" doesn't appear 
to be working, I see only 1 line displayed
- despite the seemingly permitted nat=yes entry I so far found no way to 
get this phone working from behind NAT (I know that MGCP as such probably 
will never work behind NAT, but still... by the way there is a new RFC on 
the road to attack this issue...)

NOTES:
- always restart (not reload) Asterisk after modifying mgcp.conf!
- mgcp.conf doesn't like the following keywords:
disallow=
allow=
callwaitingcallerid=
fromuser=
- the display is fine and equipped with nice logos, icons etc. The only 
thing that it is missing is the backlight, but I guess I can live with 
that...
- the menu is well organized and easily understood and navigated
- I don't think it is acceptable to only include a single A5 sheet of 
paper and not have ANY other printed user documentation shipped with 
the phone. A web-based PDF is ok, but there should be more...


Greetings,
Philipp


== hard/software info ==

Phone name Ext.987
Appli version IP10 M v0.3.0 (Build 6) 
Boot version IP10 Boot v0.3.4 
DSP version Rel 9.1.0.4, Build p8  
GG version R9.0.0 IPP (Build 5) 
IP address 192. 168. 4. 5 
Protocol MGCP 1.0 


== mgcp.conf ==

[192.168.13.5]
context=default
threewaycalling=yes
transfer=yes
callwaiting=yes
host=192.168.13.5
nat=no
canreinvite=no
callgroup=0
pickupgroup=0,1
cancallforward=yes
transfer=yes
;dtmf=inband
callerid = "John" <987>
mailbox=1234
line => aaln/1
line => aaln/2


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Re: [Asterisk-Users] Wifi600 problem

[John Todd]

At 8:49 PM +0200 11/17/03, Pertti Pikkarainen wrote:
Some of you have got Wifi600 wireless SIP phone working with Asterisk.
Specially John Todd ( nice review ).
My phones register ok. They can also receive calls from other phones.
But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ).
This seems to be due to the phone not understanding what it
should do when it receives 'Proxy Authentication Required'.
In my case it does nothing.
Can someone tell me what Wifi600 software version was used when this 
phone was succesfully tested
with Asterisk.  Any other hint is also appreciated.

--  Pertti



I had the same problem initially.  However, the vendor gave me a 
software update which fixed the authentication problem.  I had hoped 
that it would have made it out to general distribution by now. 
Please contact your vendor to see if they have the software that they 
can give to you.

If not, let me know who you're talking with and I'll see what I can 
do as far as "information transfer" to the company that sold you the 
phone so they start doing the right thing.

JT
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RE: [Asterisk-Users] dtmfmode & SIPDtmfMode

[Jordi Haarman]

Martin,

As I said in my previous email I was using a SIP phone (SJphone). If you
have a solution please let me know.

Regards,

Jordi

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Friday, November 14, 2003 10:28 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dtmfmode & SIPDtmfMode

You must be calling SIPDtmfMode on incoming calls that are not SIP
calls.
Eg: zap call that you send to SIP ... this way it doesn't work.

regards
Martin

On Fri, 14 Nov 2003, Jordi Haarman wrote:

> I get a 'Segmentation fault' now. A false mode just shows the error
> message now.
>
> Gr
>
> Jordi
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Martin
Pycko
> Sent: Friday, November 14, 2003 6:26 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] dtmfmode & SIPDtmfMode
>
> Try again ... with latest CVS.
>
> Martin
>
> On Fri, 14 Nov 2003, Jordi Haarman wrote:
>
> > Hi,
> >
> > I would like to be able to switch dtmf mode of SIP calls of local
> > clients so the server can understand them and it can also be used
when
> > connected to a remote location. I saw that there is an application
> > called SIPDtmfMode in cvs so instead of using the debian package I
> > recompiled the kernel and compiled asterisk from CVS.
> >
> > When I use the command ( exten => _XXX,1,SIPDtmfMode(inband) ) it
does
> > not seem to work. Even putting a false mode does not give me a
warning
> > or something. Did I miss something?
> >
> > Any help/suggestion is appreciated!
> >
> > gr
> >
> > Jordi
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___
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>
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[Asterisk-Users] capi config

[Rattana BIV]

Hi,
 
 
I have DIVA server BRI with 2 channels and i use 
chan_capi drivers. But I only can use 1 channel. I make one call it works, but 
if I make a second call asterisk says me = Everyone is busy at this 
time.
 
How can I configure it ?
 
 
 
Best regards
 
 
Rattana

Re: [Asterisk-Users] (no subject)

[Ryan Tucker]

On Mon, 17 Nov 2003 20:30:09 -0500 (EST), Bob Bevins <[EMAIL PROTECTED]> 
wrote:
-- Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/voicemail/local/31/INBOX/msg0011 format: wav49,
0x80de8b8
-- x=1, open writing:
/var/spool/asterisk/voicemail/local/31/INBOX/msg0011 format: gsm,
0x80f8588
-- x=2, open writing:
/var/spool/asterisk/voicemail/local/31/INBOX/msg0011 format: wav,
0x810f090
-- User hung up
  == Spawn extension (extensions, 31, 2) exited non-zero on
'[EMAIL PROTECTED]/1'
Hmm... this looks rather normal.  What seems to be the problem?  I didn't 
quite get what's actually happening out of your message... -rt

--
Ryan Tucker
Network Engineer
NetAccess, Inc.
1159 Pittsford-Victor Road
Bldg. 5, Suite 140
Pittsford, New York 14534
585-419-8200
www.netacc.net
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Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)

[Florian Overkamp]

At 09:45 18-11-2003 -0500, you wrote:
And yes, they can run fine together(I'm not using VOIP, just a T1 out of
Asterisk to Bayonne to test and see if it would work). The IVR application
that I currently still have running on Bayonne is only still on Bayonne
because it can never go down, and Bayonne has proven itself to me to be
extremely stable, while I cannot personally say AT THIS TIME that an
Asterisk box would stay up for over 6 months with no crashes.
Actually in this light it might be cute to have an 'uptime' counter inside 
asterisk (maybe a lastlog that can also show the reason of the last restart 
- was it a stop gracefully or did it just crash?) *grin*

grt,
Florian
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[Asterisk-Users] ISDN Card Types for Europe

[Ray Burkholder]

Title: ISDN Card Types for Europe






What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ?  For example:  AVM C2 or AVM C4 or eicon Diva server 4 BRI?  Any others?  Which driver is appropriate?

Ray Burkholder

[EMAIL PROTECTED]

http://www.oneunified.net

704 576 5101



-- 
Scanned for viruses & dangerous content at 
One Unified
and is believed to be clean.

[Asterisk-Users] Will Asterisk be supporting RTCP XR in the future?

[Lee Goodman]

This article below came up on the newwire. The RTCP XR RFC was published.
Will Asterisk be supporting this function in a future release? Does anyone
know if any phone vendors are going to be supporting it?

Thanks

Lee Goodman

Our Technology Update this week is about one of those

mechanisms. Known as RTP Control Protocol Reporting Extensions

(RTCP XR), the technology defines a standard way to detect VoIP

call quality by monitoring a variety of key call ingredients

such as packet loss, delay and call quality.

According to our Technology Update author

(  ), the IETF this month published

RTCP XR as RFC 3611.

In a nutshell, our author says RTCP XR works by exchanging

messages containing key call-quality-related metrics are

periodically between IP phones and gateways. This lets a probe

or analyzer monitor these metrics midstream to support problem

resolution, or be retrieved from a gateway using SNMP.

Administrators can use SNMP to retrieve data from each IP

gateway, or use midstream probes or analyzers to capture

call-quality data to aid in problem resolution.

In addition the set of VoIP performance metrics defined in RTCP

XR also form the basis for new draft quality-of-service

reporting extensions to leading call-control protocols. This

will let IP endpoints report call-quality metrics directly to

call managers and softswitches, making integration into call

detail records easier, according to our author.

I am sure you'll be hearing more about this protocol. For more

on this article see:



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Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0

[Dan]

Hi,

- Original Message - 
From: "Michael Van Donselaar" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 18, 2003 5:11 PM
Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP,
Red Hat 9.0


> On Tue, 18 Nov 2003 09:36:23 -0500, you wrote:
>
> >Tried two different Win2k systems and it crashes on load.
>
> You currently have to run it from the directory it is located in.  If you
make a
> shortcut, make sure that "Run In:" has the correct directory.
>
> iaxComm needs to see the resources in ${cwd}/rc
>
> I hope that fixes it.  Please let me know

I cannot start it.
It loads into the memory, but no window is displayed.
I have installed even wxWindows 2.4.2 framework.

Tried from command prompt, from explorer, ...nothing.

What can it be wrong?

Thanks,
Dan


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Re: [Asterisk-Users] Wifi600 problem

[Pertti Pikkarainen]

Thanks John,

Can you check what version you are using ?
I can start with the very same ( once I get it ).
I have sent a request to BCM but haven't got any reply yet.
-- Pertti





John Todd wrote:

At 8:49 PM +0200 11/17/03, Pertti Pikkarainen wrote:

Some of you have got Wifi600 wireless SIP phone working with Asterisk.
Specially John Todd ( nice review ).
My phones register ok. They can also receive calls from other phones.
But for some reason I can't make them call out ( anybody, ie. SIP or 
PSTN ).

This seems to be due to the phone not understanding what it
should do when it receives 'Proxy Authentication Required'.
In my case it does nothing.
Can someone tell me what Wifi600 software version was used when this 
phone was succesfully tested
with Asterisk.  Any other hint is also appreciated.

--  Pertti



I had the same problem initially.  However, the vendor gave me a 
software update which fixed the authentication problem.  I had hoped 
that it would have made it out to general distribution by now. Please 
contact your vendor to see if they have the software that they can 
give to you.

If not, let me know who you're talking with and I'll see what I can do 
as far as "information transfer" to the company that sold you the 
phone so they start doing the right thing.

JT
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Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0

[Michael Van Donselaar]

On Tue, 18 Nov 2003 17:02:42 +0200, "Dan" <[EMAIL PROTECTED]> wrote:

>Hi,
>
>Tried on WinXP Pro and it loads, but in the background (no window).
>There is something needed from the wxWindows package to just run the
>executable?

Nothing needed from the wxWindows package.  I think it's because it can't find
the rc directory.

I'm sorry that I didn't put this in the README.  Bad coder.  No donut.

You must run iaxComm from the installation directory beacuse it looks for rc
files in ${cwd}/rc.

Steve put an error dialog on failure in the CVS sources, but I'm working on a
better solution.

Please let me know if this solves it, or if the problem lies elsewhere.
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[Asterisk-Users] Re: ask problem about softphone--asterisk--softphone, Urgent!!!

[Qian Lv]

Hi, I want to correct an error, in my figure, the softphone2's name is Raytec, not Reltec. As the figure below shows:
Thanks!
Softphone1<>Asterisk SIP<>Softphone2(User Agent)(Proxy) (User Agent)155.69.xx.xx 155.69.yy.yy  155.69.zz.zz   zhou  mysipproxy.comRaytec
Thanks!Regards,===Lv Qian,Ph.D Student,School of Computer Engineering,Nanyang Technological University,Singapore 639798=
Do you Yahoo!?
Protect your identity with Yahoo! Mail AddressGuard

[Asterisk-Users] DIAX - Can place a call, but can't be called?!

[Todd Taylor]

Title: DIAX - Can place a call, but can't be called?!






Greetings,


DIAX seems to work well placing calls, but I can't actually receive a call . Here, DIAX (x305) "registers", then I use a sip phone to place a call to DIAX (which definitely is not in use by me at debug time, but it is idle on my desktop…I think), and then * goes to vmail. 

Here's the debug output:


affinity*CLI> iax debug

IAX Debugging Enabled

Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: REGREQ 

Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK    

Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: REGAUTH

Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: REGREQ 

Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK    

Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: ACK    

Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: REGREQ 

Tx-Frame Retry[-01] -- Seqno: 01  Type: IAX Subclass: ACK    

    -- Registered '305' (AUTHENTICATED) at 192.168.1.102:5036

Tx-Frame Retry[000] -- Seqno: 01  Type: IAX Subclass: REGACK 

Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: ACK    

    -- Executing Macro("SIP/302-b3b3", "oneline|IAX/305") in new stack

    -- Executing Dial("SIP/302-b3b3", "IAX/305|20|tr") in new stack

  == Everyone is busy at this time

    -- Executing VoiceMail("SIP/302-b3b3", "b305") in new stack

  == Parsing '/etc/asterisk/voicemail.conf':   == Parsing '/etc/asterisk/voicemail.conf': Found

    -- Playing 'vm/305/busy' (language 'en')

  == Spawn extension (macro-oneline, s, 102) exited non-zero on 'SIP/302-b3b3' in macro 'oneline'

  == Spawn extension (toll-access, s, 1) exited non-zero on 'SIP/302-b3b3'

affinity*CLI> iax no debug


What is creating the 'busy' state?  FYI, I'm not having trouble with any other FXS or SIP phones, just DIAX.  I've tried using * builds from last week and one from a couple weeks prior with no luck.

TIA…Todd

RE: [Asterisk-Users] Transfer directly to voicemail?

[Alex Nikolov Telesoft Ltd.]

Title: RE: [Asterisk-Users] Transfer directly to voicemail?





Does anyone have a setup ware a user can pickup voice messages from a group type voice box and from his own one in one go to the voice mail?

or to notify the user that hi have a message in his (sales) group mail box


Many Thanks


Alex

Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)

[Steven Critchfield]

On Tue, 2003-11-18 at 09:53, Florian Overkamp wrote:
> At 09:45 18-11-2003 -0500, you wrote:
> >And yes, they can run fine together(I'm not using VOIP, just a T1 out of
> >Asterisk to Bayonne to test and see if it would work). The IVR application
> >that I currently still have running on Bayonne is only still on Bayonne
> >because it can never go down, and Bayonne has proven itself to me to be
> >extremely stable, while I cannot personally say AT THIS TIME that an
> >Asterisk box would stay up for over 6 months with no crashes.
> 
> Actually in this light it might be cute to have an 'uptime' counter inside 
> asterisk (maybe a lastlog that can also show the reason of the last restart 
> - was it a stop gracefully or did it just crash?) *grin*

Hmm, maybe it wouldn't be much of a hack to get at the show uptime
information and dump it with each log as it is sent to the events.log
file so you can see if certain length runtimes cause crashes as well.
Especially with respect to the post I just read about the user who has
asterisk "going nuts every day".

I wouldn't be opposed to it being put in the CLI prompt too, or maybe
just made available and then we could do something like the PS1
formatting of the prompt.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)

[Jared Smith]

On Tue, 2003-11-18 at 08:53, Florian Overkamp wrote:
> Actually in this light it might be cute to have an 'uptime' counter inside 
> asterisk (maybe a lastlog that can also show the reason of the last restart 
> - was it a stop gracefully or did it just crash?) *grin*

Have you tried "show uptime" from the Asterisk CLI?

Jared Smith

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[Asterisk-Users] telco access ?s -- PRI, T1, POTS?

[john lawler]

Hi guys,

I'm new to the telco game and still pretty new to Asterisk, although 
I've been using it for a couple of months now and like most of what I 
see.  At my office, we've got a small two extension setup w/ two Digium 
cards for a single FXO line and three FXS extensions, but I'm also 
working on designing a larger installation for a customer which will 
involve ~16 analog handsets that I'll be running through a Rhino 
Equipment channel bank to a Digium T1.  My question is all about what 
type of phone service would be recommended for my local provider.

I purchased a CAC FXO channel bank to interface with a second Digium T1 
card that I've got, and ran some small scale tests on that which seemed 
to go fine, and I'm sure this solution would be *sufficient* for the 
customer, but I know he would like to be able to do DID, which, from 
conversations w/ SBC in my area, I've discovered requires something 
other than POTS.  We'll be looking to have roughly 10-12 phone lines for 
both inbound and outbound (as you'd have in the POTS world), or w/e the 
PRI, etc. equivalent would be.

So the real question is, from all of your experience, assuming my local 
telco can provide me w/ any possible solution, what would you all 
recommend I choose?  Some sort of fractional T1 line w/ DID trunks for 
incoming and POTS for outbound calls (pardon my description of these 
solutions if they are inaccurate--again, I'm new to the game) or what 
else? 

The other main point I should make is that both the customer and myself 
are interested in a solution where we could, in an emergency, or in the 
event my configuration of Asterisk is inadequate, switch back over to a 
situation where we could at least place and receive all calls using the 
analog handsets.  (E.g., from what I know about this stuff, I suppose I 
could plug the Rhino FXS channel bank into an incoming T1 line which 
would then split all the lines into analog which my phones could then 
plug right into--of course, if I went w/ a straight POTS solution, we 
could obviously just plug 10-12 of the analog handsets into the lines 
and at least answer the phones (albeit in a haphazard manner)).

I apologize for the length of this question, but I figured some of the 
knowledgeable people on this list would've had some experience in these 
areas.

Thanks,

jl
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Re: [Asterisk-Users] I hate to do this but..

[Jason A. Pattie]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason A. Pattie wrote:
| I have this exact same problem as well.  We have a scenario in which we
| are not using any analog extensions, just SIP and IAX software based
| phones (DIAX, X-Lite, gnophone, (trying to use) linphone, etc.) with a
| single incoming POTS line to an X101P card.  First, the audio level
| coming from/to the X101P card is unacceptable, unless rxgain/txgain are
| turned up.  It starts becoming usable around 6.0.  The desired audio
| level is reached when the gains are increased to between 10.0 and 13.0.
| At 13.0, though, the sound becomes almost too distorted to be of use,
| but it is at a very acceptable loudness level.
|
| As far as echo is concerned, there don't seem to be any problems on the
| side of the person calling into the system to the X101P card.  However,
| if they connect to a SIP/IAX extension, or the SIP/IAX extension makes
| an outgoing call through the X101P card, the echo on the SIP/IAX
| extension is horrendous.  Of all the echo suppressors I've tried, MARK3
| works the best for not blowing your eardrums out.  The problem is
| excrutiating when using MARK2 with AGGRESSIVE cancellation.  When this
| is enabled, and the rx/txgains are anywhere above 0.0 (even 0.5), the
| SIP/IAX client receives some sort of feedback or distortion that can
| only be described as an extremely loud screech or squelch that hurts
| very much.  And, it is rather explosive in nature.  It is partly solved
| by turning down the microphone volume and gain on the SIP/IAX client,
| but it does not stop the problem.  If the rx/txgains are set to 0.0,
| then the volume level coming from/through? the X101P is so low, that
| even turning the volume up all the way on the SIP/IAX client, it is
| extremely difficult to make out what the other party is saying (there is
| a bit of ambient sound level in the location we are working from,
| though).  The same happens if voicemail is left using the X101P.
| However, if voicemail is left using the SIP/IAX client, volume levels
| are very acceptable.
|
| I don't understand what is going wrong.  Do I have a bad X101P card?
| I've tried this card in 3 different machines, so far.  I will be trying
| it in a fourth shortly and let you know how it goes.
Well, I've now tried it in my fourth machine, a PII 266MHz, 128MB RAM
system.  It's slightly better, and I have tried it using MARK3 and MARK2
with AGGRESSIVE enabled.  It seems that it doesn't do bad things when
the rx/txgain is set to 3.0 with MARK2 and AGGRESSIVE, but there is
still a slight echo of the speaker of the VOIP extension (SIP/IAX).  The
echo isn't as bad, because every time they talk, the beginning of the
echo starts fine, but the end gets chopped off, so it's kind of a
stuttering echo, now, which is more usable than before.  I've also tried
multiple combinations of echocancel on and off, and
echocancelwhenbridged on and off as well as the full range of echocancel
being set to 32, 64, 128, and 256.  128 seems to work the best with 256
making the echo seeming to be slightly louder and more distorted.  The
audio input now seems to be at a reasonable level when listening to the
phone line, but it is still a bit soft.  However, the voicemail
recording is still so low as to be almost unintelligible.
Thanks.

- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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RE: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Philipp von Klitzing]

Hi!

> http://www.nokia.com/nokia/0,8764,43170,00.html
> 
> Two PSTN/GSM gateways called the Nokia 22 and the Nokia 32. I don't
> know how it will operate with * yet but it seems rather transparant. 

Interesting. You could also use a "Siemens M20 Terminal" (I still have 
one around here), or look at some slightler newer product of the same 
type, e.g. http://www.tdc.co.uk/gsm/.

Next to that you could also try any GSM PCcard (PCMCIA).

I wonder, however, if chan_modem will be sufficient for this.

Cheers, Philipp


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Re: [Asterisk-Users] VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk

[Stephen R. Besch]

On Mon, 2003-11-17 at 08:34, Steve Murphy wrote:
 

Hello--

I've been asked an interesting question, and I'm too ignorant to answer
it authoritatively (yet). Can anyone help me?
Question: If I'm going to implement a somewhat small (10-80) phone
system, and I have a choice of using VOIP phoneset (like SNOM or
Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly
what features will I kiss goodbye if I use the cheap analogs?
In other words, what features will a (more expensive) VOIP phoneset
provide, that the analog won't?
   

If you need any off-site extensions - I have one in Nova Scotia off our 
Buffalo server - then VoIP is great, and dramatically cheaper than 
analog.  However, there is nothing stopping you using a mixed 
environment. Put analog phones where the wiring is easy and VoIP where 
the wiring is hard and/or there is already suitable ethernet infrastructure.

Stephen R. Besch

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Re: [Asterisk-Users] Wifi600 problem

[John Todd]

The version of the WiFi software that I am running that is confirmed 
to work with Asterisk is wb000_d.img

JT


Thanks John,

Can you check what version you are using ?
I can start with the very same ( once I get it ).
I have sent a request to BCM but haven't got any reply yet.
-- Pertti

John Todd wrote:

At 8:49 PM +0200 11/17/03, Pertti Pikkarainen wrote:

Some of you have got Wifi600 wireless SIP phone working with Asterisk.
Specially John Todd ( nice review ).
My phones register ok. They can also receive calls from other phones.
But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ).
This seems to be due to the phone not understanding what it
should do when it receives 'Proxy Authentication Required'.
In my case it does nothing.
Can someone tell me what Wifi600 software version was used when 
this phone was succesfully tested
with Asterisk.  Any other hint is also appreciated.

--  Pertti



I had the same problem initially.  However, the vendor gave me a 
software update which fixed the authentication problem.  I had 
hoped that it would have made it out to general distribution by 
now. Please contact your vendor to see if they have the software 
that they can give to you.

If not, let me know who you're talking with and I'll see what I can 
do as far as "information transfer" to the company that sold you 
the phone so they start doing the right thing.

JT
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Re: [Asterisk-Users] help voicepulse connect

[Azher Amin]

ohhh , thnx. btw it was my typo mistake :)
 
AzherBrian West <[EMAIL PROTECTED]> wrote:
Hey dude... they email you the config.. but you might wanna have yourpriority numbers correct.exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},40,tT)exten => _1NXXNXX,2,Playback,vm-goodbyeOn Mon, 17 Nov 2003, Azher Amin wrote:>> voicepulse works fine for me ..>> In extensions.conf>> for usa dialing> exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},40,tT)> exten => _1NXXNXX,1,Playback,vm-goodbye>> for international dialing> exten => _1011.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN-1}> exten => _1011.,1,Playback,vm-goodbye>> In iax.conf>> [voicepulse]> context = foo> secret=secret> auth=md5> type=friend> host=gw5.voicepulse.com>> I hope it
  will
 work for you as well.>> Azher>>> listas iPfone <[EMAIL PROTECTED]>wrote:> Hi All>> I signed up for an account with voicepulse connect service and received the info to set up asterisk.>> Anyone have that confs to send as an example?>> Thanks>> Miklos -> Do you Yahoo!?> Protect your identity with Yahoo! Mail AddressGuard___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] telco access ?s -- PRI, T1, POTS?

[Steven Critchfield]

On Tue, 2003-11-18 at 11:04, john lawler wrote:
> Hi guys,
> working on designing a larger installation for a customer which will 
> involve ~16 analog handsets that I'll be running through a Rhino 
> Equipment channel bank to a Digium T1.  My question is all about what 
> type of phone service would be recommended for my local provider.
> 
> I purchased a CAC FXO channel bank to interface with a second Digium T1 
> card that I've got, and ran some small scale tests on that which seemed 
> to go fine, and I'm sure this solution would be *sufficient* for the 
> customer, but I know he would like to be able to do DID, which, from 
> conversations w/ SBC in my area, I've discovered requires something 
> other than POTS.  We'll be looking to have roughly 10-12 phone lines for 
> both inbound and outbound (as you'd have in the POTS world), or w/e the 
> PRI, etc. equivalent would be.

> The other main point I should make is that both the customer and myself 
> are interested in a solution where we could, in an emergency, or in the 
> event my configuration of Asterisk is inadequate, switch back over to a 
> situation where we could at least place and receive all calls using the 
> analog handsets.  (E.g., from what I know about this stuff, I suppose I 
> could plug the Rhino FXS channel bank into an incoming T1 line which 
> would then split all the lines into analog which my phones could then 
> plug right into--of course, if I went w/ a straight POTS solution, we 
> could obviously just plug 10-12 of the analog handsets into the lines 
> and at least answer the phones (albeit in a haphazard manner)).

Your wish for DID and the wish to remove asterisk from the loop in case
of failure are going to clash with each other. DID can be delivered via
E&M wink, but this will require a machine on your end that will work
with that signaling. E&M wink basically is like having the telco dial an
extension in your PBX for your caller. 

PRI would not allow you to just swap out to analog phones at all as the
Rhino will not understand the signaling on the 24th channel.

You would be best to get very familiar with this setup, and go with the
frac T1. If you don't feel comfortable deploying yet, then you aren't
ready to deploy. I don't wish to seem mean here, but this is a critical
component of your clients business life and if you can't pull it off,
then you don't have any business experimenting on your customers money.
Read the archives of a similar fellow who had a asterisk system not work
out for his client. 

Right now, I'd say a fair knowledge of the internals of asterisk should
be a requirement for anyone selling asterisk systems into a client site.
This would help you go far in justifying your knowledge level to your
customer. If you could even point to your name in the credits file, this
would be a very good selling point to your clients. If you don't want to
get dirty in the coding, thats fine, but you should be able to debug
sections of the code when you find a problem.

So far I know there are several vendors on this list that have their
names in the credits file, and are active here. If you don't feel you
can go it alone, you would also do well to find one of these vendors and
see if they can be contracted out on a contingency basis to cover what
you don't know.

BTW, I am not a vendor. 

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[John Todd]

At 1:20 PM + 11/18/03, Linus Surguy wrote:
 > I voulenteer with a local Search And Rescue unit and I was speaking with
the
 > senior members about how they interface the command trailer PBX with the
PSTN
 > or cellular networks when they are on scene at a remote location.  Turns
out
 > they don't.  Thus that got me to thinking about how one would get
Asterisk to
 > interface with a cell phone directly, or what hardware out there works
well
 > for the task.  I have managed to figure out how to make my Motorola
StarTAC
 > make outgoing voice calls for instance and making an interface for this
phone
 > should be quite possible, but it would be nice to have incoming phone
calls
 > too.  I know there are many companies out there which make such producs,
and
 > that a Google search can yield many results, I want thoughts and
opinions,
 > stuff which Google suks at.

 Some of the older cell phones use to expose either a 2-wire or 4-wire
 interface via a connector on the phone (don't know about transmission
I won't bother with any of that - purchase a Nokia Premicell (or other
manufacturers similar item). This device takes a normal GSM SIM card and
then presents a normal PSTN line interface - plug that into your normal
Asterisk PSTN line card - job done.
Linus
I don't have time to go digging for the link, but there is at least 
one vendor who offers a hardware solution with 24 DS0 GSM cards in a 
channel bank, which you'd then connect to a T100P card.  In other 
words, it's 24 lines of cell phones in a 2u package, with minimal 
wiring hassle.  Google should show you the way...

JT
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RE: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Thilo Salmon]

> I have a similar challenge. I will have to connect a remote location with
> PBX to a central location with PBX. While roaming the Internet I came
> accross this:

If this location is in the netherlands, a look at
http://www.falcom.de/e/e-produkte.html might also prove helpful. They
offer a broad array of embedded GSM/GPRS/GPS modules. I used a number of
A2D devices for a messaging application. They operate almost like a
hayes compatible voice modem over a serial link. Only they come with a
proprietary command set to do GSM related tasks such as entering a PIN
number for your SIM.

Thilo

P.S.: I still have a number of them lying around. Drop me an email, if
you need some.

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[Asterisk-Users] Asterisk with External Voicemail

[cveazey]

If anyone could help me with this, I'd appreciate it!  

I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system.  I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number.  That voicemail system works by seeing the CALLED number and routing the call to the user's answering message.  The problem is that Asterisk sends the CALLING digits on the outbound call and the CALLING party gets the "welcome to Octel" message instead of the user's voicemail box.

Is there any way to a variable like ${EXTEN} or ${RDNIS} to "replace" the caller ID information with the destination mail box number?  

Am I thinking through this the wrong way?

Thanks!

Chris Veazey

RE: [Asterisk-Users] ISDN Card Types for Europe

[tan]

Title: Message



We 
deploy the Eicon Diva Server range of cards for production systems as they have 
onboard echo cancellation, and work very well with chan_capi and 
asterisk.
 
Tan
www.voiptalk.org
 
 

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Ray 
  BurkholderSent: 18 November 2003 16:01To: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] ISDN Card 
  Types for Europe
  What types of ISDN BRI cards work well in Europe 
  (Guadeloupe, Martinique and France) 
  ?  For example:  AVM C2 or AVM C4 or eicon 
  Diva server 4 BRI?  Any others?  Which driver is 
  appropriate?
  Ray Burkholder 
  [EMAIL PROTECTED] 
  http://www.oneunified.net 
  704 576 5101 
  -- Scanned for viruses & dangerous content at One Unified and is believed to be clean. 

Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[TC]

>interface the command trailer PBX with the PSTN
> or cellular networks when they are on scene at a remote location.  Turns
out
> they don't.  Thus that got me to thinking about how one would get Asterisk
to
> interface with a cell phone directly, or what hardware out there works
well
> for the task.
These guys have a wide range of cellular adapters
http://www.telular.com/products/index.asp

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Re: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network

[Linus Surguy]

> >>
> >>  Some of the older cell phones use to expose either a 2-wire or 4-wire
> >>  interface via a connector on the phone (don't know about transmission
> >
> >I won't bother with any of that - purchase a Nokia Premicell (or other
> >manufacturers similar item). This device takes a normal GSM SIM card and
> >then presents a normal PSTN line interface - plug that into your normal
> >Asterisk PSTN line card - job done.
>
> I don't have time to go digging for the link, but there is at least
> one vendor who offers a hardware solution with 24 DS0 GSM cards in a
> channel bank, which you'd then connect to a T100P card.  In other
> words, it's 24 lines of cell phones in a 2u package, with minimal
> wiring hassle.  Google should show you the way...

Indeed there are - we looked at one of these for a different project, here
is some details from a German vendor:

http://www.teles-communication-systems.com/G1/G19500AN0_GSM_frame.html

Linus

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[Asterisk-Users] Help with Warnings

[Andy Hester]

I'm trying to clean up some notices/warnings that are repeatedly logged
in *.Any Help would be appreciated as I'm not sure of the cause
/solution.

Here are the errors:

Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321
(zt_call): cidspill already exists??

+
/* Don't send audio while on hook, until the call is answered */
p->dialing = 1;
if (p->use_callerid) {
/* Generate the Caller-ID spill if desired
*/
if (p->cidspill) {
ast_log(LOG_WARNING, "cidspill
already exists??\n");
free(p->cidspill);
}
++



Nov 17 17:13:51 NOTICE[1242768320]: File app_dial.c, Line 502
(dial_exec): Unable to create channel of type 'Zap'

++
/* Request the peer */
tmp->chan = ast_request(tech, chan->nativeformats,
numsubst);
if (!tmp->chan) {
/* If we can't, just go on to the next call */
ast_log(LOG_NOTICE, "Unable to create channel of
type '%s'\n", tech);
if (chan->cdr)
ast_cdr_busy(chan->cdr);
free(tmp);
cur = rest;
continue;
}
++



Nov 17 17:20:57 NOTICE[1209214400]: File chan_zap.c, Line 3462
(zt_read): Fax detected, but no fax extension

++
/* Fax tone -- Handle and return NULL */
if (!p->faxhandled) {
p->faxhandled++;
if (strcmp(ast->exten, "fax")) {
if (ast_exists_extension(ast,
ast->context, "fax", 1,
ast->callerid)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_
PREFIX_3 "Redirecting %s to fax extension\n",
ast->name);
/* Save the DID/DNIS when we
transfer the fax call to a "fax"
extension */
pbx_builtin_setvar_helper(as
t,"FAXEXTEN",ast->exten);
if (ast_async_goto(ast,
ast->context, "fax", 1, 0))
ast_log(LOG_WARNING,
"Failed to async goto '%s' into fax of
'%s'\n", ast->name, ast->context);
} else
ast_log(LOG_NOTICE, "Fax
detected, but no fax extension\n");
} else
++



Nov 17 17:26:04 WARNING[1234379584]: File chan_zap.c, Line 3331
(zt_read): zt_rec: Unknown error 500

+++
/* Check for hangup */
if (res < 0) {
if (res == -1)  {
if (errno == EAGAIN) {
/* Return "NULL" frame if there is nobody
there */
ast_mutex_unlock(&p->lock);
return &p->subs[index].f;
} else
ast_log(LOG_WARNING, "zt_rec: %s\n",
strerror(errno));
}
ast_mutex_unlock(&p->lock);
return NULL;
+


thanks,
Andy

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Re: [Asterisk-Users] DIAX - Can place a call, but can't be called?!

[Dan]

Hi,

- Original Message - 
From: "Todd Taylor" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 18, 2003 6:22 PM
Subject: [Asterisk-Users] DIAX - Can place a call, but can't be called?!


> Greetings,
>
> DIAX seems to work well placing calls, but I can't actually receive a
> call . Here, DIAX (x305) "registers", then I use a sip phone to place a
> call to DIAX (which definitely is not in use by me at debug time, but it
> is idle on my desktop.I think), and then * goes to vmail.
>
> Here's the debug output:
>
> affinity*CLI> iax debug
> IAX Debugging Enabled
> Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: REGREQ
> Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK
> Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: REGAUTH
> Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: REGREQ
> Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK
> Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: ACK
> Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: REGREQ
> Tx-Frame Retry[-01] -- Seqno: 01  Type: IAX Subclass: ACK
> -- Registered '305' (AUTHENTICATED) at 192.168.1.102:5036
> Tx-Frame Retry[000] -- Seqno: 01  Type: IAX Subclass: REGACK
> Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: ACK
> -- Executing Macro("SIP/302-b3b3", "oneline|IAX/305") in new stack
> -- Executing Dial("SIP/302-b3b3", "IAX/305|20|tr") in new stack
>   == Everyone is busy at this time
> -- Executing VoiceMail("SIP/302-b3b3", "b305") in new stack
>   == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
> '/etc/asterisk/voicemail.conf': Found
> -- Playing 'vm/305/busy' (language 'en')
>   == Spawn extension (macro-oneline, s, 102) exited non-zero on
> 'SIP/302-b3b3' in macro 'oneline'
>   == Spawn extension (toll-access, s, 1) exited non-zero on
> 'SIP/302-b3b3'
> affinity*CLI> iax no debug
>
> What is creating the 'busy' state?  FYI, I'm not having trouble with any
> other FXS or SIP phones, just DIAX.  I've tried using * builds from last
> week and one from a couple weeks prior with no luck.

Very strange
There is any (personal)firewall on the pc running DIAX (or if the OS is XP
Pro, it is activated the internal one)?
If yes, take care to open the UDP port 5036..

Best regards,
Dan


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Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0

[Kevin Bockman]

>--- "Andrew Thompson" <[EMAIL PROTECTED]> wrote:
>Tried two different Win2k systems and it crashes on load.

Doesn't crash for me, just don't get anything.  Continues to run in the background but 
no interface.  I have to ctrl-alt-del to end it.

Kevin



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Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0

[Kevin Bockman]

>Tried two different Win2k systems and it crashes on load.

>You currently have to run it from the directory it is located in.  If >you make a
>shortcut, make sure that "Run In:" has the correct directory.

>iaxComm needs to see the resources in ${cwd}/rc

Ok, I tried 'running it from the directory it is located in', and it does crash for 
me.  If I start it from Start, Run -- it just sits there no GUI.  I have to 
ctrl-alt-del to kill it from the background processes.

Kevin

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Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0

[Dan]

Hi,

- Original Message - 
From: "Kevin Bockman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 18, 2003 8:26 PM
Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP,
Red Hat 9.0


> >--- "Andrew Thompson" <[EMAIL PROTECTED]> wrote:
> >Tried two different Win2k systems and it crashes on load.
>
> Doesn't crash for me, just don't get anything.  Continues to run in the
background but no interface.  I have to ctrl-alt-del to end it.

The same for me...

Dan


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[Asterisk-Users] manager.conf

[George Lin]

Hi,

 Do you know if we can use AGI or other script to handle the
 asterisk events by using the existing asterisk manager process ?

 Please advise.

 Thanks

 George


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[Asterisk-Users] Re: chan_zap won't load after CVS update

[Matt Lawson]

Ah ha.   That's *almost* got it.  It will now load and * will run.  The 
only big gotcha is it won't pick up or dial out on a POTS line.  ztcfg 
shows both channels configured OK, as does 'zap show channels.'  If I 
try to dial out I get:

  -- Executing Goto("SIP/3063-74d0", "outside|9555|1") in new 
stack  

-- Goto 
(outside,9555,1)   

-- Executing Dial("SIP/3063-74d0", "Zap/g1/ww954...") in new 
stack
== Everyone is busy at this time

I had the same problem the other day. Resolved it by essentially
blowing away the existing src directories (rename them if you want) and
doing a new cvs checkout. I know I seen someone post something a month
or two ago relative to using a cvs flag to effectively over-write
everything on the local system. Guess we're all supposed to be linux
cvs experts.
In my case, I was simply trying to do an update to asterisk (without
zaptel, zapata, etc), and that didn't work since some constants were
apparently defined in some non-asterisk header file (zaptel, I think, 
but I didn't bother to search it out) that were needed in one of the
channels modules. Sounds like you've bumped against the same issue.

 

Regards Mick West



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Re: [Asterisk-Users] Radius on *

[Olle E. Johansson]

SIP Express Router have radius support. Look there for hints on how to get
Radius support for VOIP.
http://iptel.org/ser/

/O

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[Asterisk-Users] Musisc on hold insted of Ringing tone

[Bartosz Jozwiak]

 
How to set up MusicOnHold insted of Ringng 
tone?
 
Bart.

Re: [Asterisk-Users] Asterisk with External Voicemail

[Steven Critchfield]

On Tue, 2003-11-18 at 11:56, [EMAIL PROTECTED] wrote:
> If anyone could help me with this, I'd appreciate it!  
> 
> I've got an Asterisk deployment where I'd like to use an existing
> external Octel voicemail system.  I've been trying to define an
> extension that if the call isn't answered in a few rings, to dial our
> external voicemail number.  That voicemail system works by seeing the
> CALLED number and routing the call to the user's answering message.
> The problem is that Asterisk sends the CALLING digits on the outbound
> call and the CALLING party gets the "welcome to Octel" message instead
> of the user's voicemail box.
> 
> Is there any way to a variable like ${EXTEN} or ${RDNIS} to "replace"
> the caller ID information with the destination mail box number?  
> 
> Am I thinking through this the wrong way?

-- show application SetCallerID  --

  -= Info about application 'SetCallerID' =- 

[Synopsis]:
  Set CallerID

[Description]:
  SetCallerID(clid[|a]): Set Caller*ID on a call to a new
value.  Sets ANI as well if a flag is used.  Always returns 0

-- show application SetCIDName  --

  -= Info about application 'SetCIDName' =- 

[Synopsis]:
  Set CallerID Name

[Description]:
  SetCIDName(cname[|a]): Set Caller*ID Name on a call to a new
value, while preserving the original Caller*ID number.  This is
useful for providing additional information to the called
party. Sets ANI as well if a flag is used.  Always returns 0

-- show application SetCIDNum  --

  -= Info about application 'SetCIDNum' =- 

[Synopsis]:
  Set CallerID Number

[Description]:
  SetCIDNum(cnum[|a]): Set Caller*ID Number on a call to a new
value, while preserving the original Caller*ID name.  This is
useful for providing additional information to the called
party. Sets ANI as well if a flag is used.  Always returns 0


-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Help with Warnings

[Steven Critchfield]

Smells like a -dev type discussion. 

Some of these messages could be removed for your use, or just change
what you want to see in your logs.


On Tue, 2003-11-18 at 12:18, Andy Hester wrote:
> I'm trying to clean up some notices/warnings that are repeatedly logged
> in *.Any Help would be appreciated as I'm not sure of the cause
> /solution.
> 
> Here are the errors:
> 
> Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321
> (zt_call): cidspill already exists??
> 
> +
> /* Don't send audio while on hook, until the call is answered */
> p->dialing = 1;
> if (p->use_callerid) {
> /* Generate the Caller-ID spill if desired
> */
> if (p->cidspill) {
> ast_log(LOG_WARNING, "cidspill
> already exists??\n");
> free(p->cidspill);
> }
> ++
> 
> 
> 
> Nov 17 17:13:51 NOTICE[1242768320]: File app_dial.c, Line 502
> (dial_exec): Unable to create channel of type 'Zap'
> 
> ++
> /* Request the peer */
> tmp->chan = ast_request(tech, chan->nativeformats,
> numsubst);
> if (!tmp->chan) {
> /* If we can't, just go on to the next call */
> ast_log(LOG_NOTICE, "Unable to create channel of
> type '%s'\n", tech);
> if (chan->cdr)
> ast_cdr_busy(chan->cdr);
> free(tmp);
> cur = rest;
> continue;
> }
> ++
> 
> 
> 
> Nov 17 17:20:57 NOTICE[1209214400]: File chan_zap.c, Line 3462
> (zt_read): Fax detected, but no fax extension
> 
> ++
> /* Fax tone -- Handle and return NULL */
> if (!p->faxhandled) {
> p->faxhandled++;
> if (strcmp(ast->exten, "fax")) {
> if (ast_exists_extension(ast,
> ast->context, "fax", 1,
> ast->callerid)) {
> if (option_verbose > 2)
> ast_verbose(VERBOSE_
> PREFIX_3 "Redirecting %s to fax extension\n",
> ast->name);
> /* Save the DID/DNIS when we
> transfer the fax call to a "fax"
> extension */
> pbx_builtin_setvar_helper(as
> t,"FAXEXTEN",ast->exten);
> if (ast_async_goto(ast,
> ast->context, "fax", 1, 0))
> ast_log(LOG_WARNING,
> "Failed to async goto '%s' into fax of
> '%s'\n", ast->name, ast->context);
> } else
> ast_log(LOG_NOTICE, "Fax
> detected, but no fax extension\n");
> } else
> ++
> 
> 
> 
> Nov 17 17:26:04 WARNING[1234379584]: File chan_zap.c, Line 3331
> (zt_read): zt_rec: Unknown error 500
> 
> +++
> /* Check for hangup */
> if (res < 0) {
> if (res == -1)  {
> if (errno == EAGAIN) {
> /* Return "NULL" frame if there is nobody
> there */
> ast_mutex_unlock(&p->lock);
> return &p->subs[index].f;
> } else
> ast_log(LOG_WARNING, "zt_rec: %s\n",
> strerror(errno));
> }
> ast_mutex_unlock(&p->lock);
> return NULL;
> +
> 
> 
> thanks,
> Andy
> 
> ___
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-- 
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Re: [Asterisk-Users] Radius on *

[Rainer Jochem]

> No and there is absolutely no need for it to.   RADIUS is not anything 
> that should have ever been deployed in a VoIP environment.  


I have no need for H.323 or Skinny or SIP; IAX works fine...
H.323 is something that should never ever have been invented...
 


I fully agree with the things other people have already mentioned
on this topic. In (I think not only) my opinion database support
is something which would be very handy in several environments.
   I really like the "swiss-army knife"-like features of asterisk
and having database support for various types of db would be
a big step forward as it would help giving asterisk again more 
potential users and thus more success and public interest.

Imagine having an include-statement in e.g. your sip.conf 
like 
 db_lookup => (db-type, user:[EMAIL PROTECTED], table, ...)
and then being able to serve several thousand users more.
Could be very useful and should not harm the current design
that much.



Best regards, 

 Rainer

-- 
http://graphics.cs.uni-sb.de/~rainer/

pgp0.pgp
Description: PGP signature

Re: [Asterisk-Users] manager.conf

[Steven Critchfield]

On Tue, 2003-11-18 at 12:37, George Lin wrote:
> Hi,
> 
>  Do you know if we can use AGI or other script to handle the
>  asterisk events by using the existing asterisk manager process ?

AGI is for handling calls. AGI is to phone calls like CGI is to web page
requests.

There is a perl module to use in accessing manager events though. search
the archive for links to it. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] SIP calls no longer work

[jerk face]

I guess I should have been more specific.  When I said
"Running 'sip debug' does not solve this problem". 
What I meant was that SIP debug doesn't show me
anything.

It turns out it was a dumb codec error on my part.
Problem solved.


--- Andrew Thompson <[EMAIL PROTECTED]> wrote:
> - Original Message - 
> From: "jerk face" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, November 17, 2003 3:53 PM
> Subject: [Asterisk-Users] SIP calls no longer work
> 
> 
> > Hello,
> > I'm having a problem with SIP.  More specifically,
> I
> > can't make any calls using SIP.
> > I have had an iConnectHere account and Free World
> > Dialup account working for quite some time, and
> now
> > all of a sudden I can't make any SIP outgoing
> calls.
> > 
> > PBX*CLI> sip show registry
> > Host  Username Refresh State
> > 192.246.69.223:5060   X120
> Registered
> > 213.137.73.178:5060    120
> Registered
> > 213.137.73.178:5060    120
> Registered
> > 
> > The above CLI output shows that I am registered to
> FWD
> > and my iConnectHere accounts.
> > 
> > The following output shows what I see when I make
> a
> > call using FWD or iConnectHere:
> > Starting simple switch on 'Zap/4-1'
> > -- Executing SetCallerID("Zap/4-1", "X")
> in
> > new stack
> > -- Executing SetCIDName("Zap/4-1", ""XX"")
> in
> > new stack
> > -- Executing Dial("Zap/4-1", "SIP/[EMAIL PROTECTED]") in
> new
> > stack
> >   == Everyone is busy at this time
> > -- Hungup 'Zap/4-1'
> > 
> > 
> > Running 'sip debug' does not solve this problem.  
> 
> Running 'sip debug' is to allow you to determine the
> problem, or capture the info necessary for someone
> onlist to determine the problem. Turning it on
> doesn't solve problems by itself(not on purpose,
> anyway).
> 
> Now that you've turned it on, attempt a call, and
> paste in the messages for us.
> 
> > 
> > I'm out of ideas, has this happened to anybody
> before?
> > 
> > Thank you for your time.


=
Asterisk is my lover, and IAX2 is our scented lubricant

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[Asterisk-Users] Asterisk GUI Client Released!!!

[mattf]

Hello,

I have finished my basic polishing of the Asterisk GUI client I have been
writing in Perl/TK and have released a first beta version on sourceforge:

http://sourceforge.net/projects/astguiclient/

I am still working on a user manual for the application, but the code works
and we have been using the same basic client for the last month here at my
company and it is working just fine.

I'm eager to hear what you all have to say about it, so send me those
comments.


Here are the screen shots of the same application running on Linux and
Windows:

http://www.freedomphones.net/astguiclient_linux.gif

http://www.freedomphones.net/astguiclient_windows.gif

MATT---

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Re: [Asterisk-Users] Radius on *

[Andrew Kohlsmith]

> 
> I have no need for H.323 or Skinny or SIP; IAX works fine...
> H.323 is something that should never ever have been invented...
> 

I feel the same way about SIP.  What a nasty protocol.  :-(

Regards,
Andrew
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[Asterisk-Users] SIP Context from domain?

[Tristan 'Minty' Colgate]

Hi,

  Is it possible to pick the context of a call from chan_sip based on the
domain of the To: header of the INVUTE? I've had a quick look throught he code
and can't see anything, I want to use the voicemail virtual hosting with
chan_sip. Can the sip domain be picked out with a global in extensions.conf?
This woud also solve my problem.

  If not is there any specifc reason/restriction that I am missing? If it is
not already supported and there aren't any specific objections then I don't
mind putting together a patch for it.

  I'm working with the last stable release and haven;t checked out CVS yet.

-- 
Tristan 'Minty' Colgate
<[EMAIL PROTECTED]> | ICQ #154577755
---
  "I don't mean to sound bitter, cold, or cruel, but
 I am, so that's how it comes out"
- Bill Hicks
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Re: [Asterisk-Users] Radius on *

[Brancaleoni Matteo]

hi.

Il mar, 2003-11-18 alle 20:33, Andrew Kohlsmith ha scritto:
> > 
> > I have no need for H.323 or Skinny or SIP; IAX works fine...
> > H.323 is something that should never ever have been invented...
> > 
> 
> I feel the same way about SIP.  What a nasty protocol.  :-(

the really big problem is that h323 was designed by old-pstn
engineers and not by network (intended as data networks) engineers,
so h323 is very hard to understand, slow, network unfriendly and so on.


don't want to start a voip protocol war.
only my 2 cents here


-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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Re: [Asterisk-Users] Asterisk GUI Client Released!!!

[Brian Capouch]

mattf wrote:
Hello,

I have finished my basic polishing of the Asterisk GUI client I have been
writing in Perl/TK and have released a first beta version on sourceforge:
http://sourceforge.net/projects/astguiclient/

I am still working on a user manual for the application, but the code works
and we have been using the same basic client for the last month here at my
company and it is working just fine.
I'm eager to hear what you all have to say about it, so send me those
comments.

I wonder if there is hope for those of us who use Postgres and not MySQL?

I am anxious to run it, but probably not going to switch backend DB 
servers in order to do so.  I suspect there will be others with the same 
attitude.

Thx.

B.

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Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)

[Sri]

/var/log/asterisk/event_log should have the last time the server restarted.
This should closely match "show uptime" result.

Jared Smith wrote:

  On Tue, 2003-11-18 at 08:53, Florian Overkamp wrote:
  
  
Actually in this light it might be cute to have an 'uptime' counter inside 
asterisk (maybe a lastlog that can also show the reason of the last restart 
- was it a stop gracefully or did it just crash?) *grin*

  
  
Have you tried "show uptime" from the Asterisk CLI?

Jared Smith

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Re: [Asterisk-Users] SIP Context from domain?

[John Todd]

At 8:00 PM + 11/18/03, Tristan 'Minty' Colgate wrote:
From: "Tristan 'Minty' Colgate" <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP Context from domain?
Reply-To: [EMAIL PROTECTED]
Date: Tue, 18 Nov 2003 20:00:55 +
Hi,

  Is it possible to pick the context of a call from chan_sip based on the
domain of the To: header of the INVUTE? I've had a quick look throught he code
and can't see anything, I want to use the voicemail virtual hosting with
chan_sip. Can the sip domain be picked out with a global in extensions.conf?
This woud also solve my problem.
  If not is there any specifc reason/restriction that I am missing? If it is
not already supported and there aren't any specific objections then I don't
mind putting together a patch for it.
  I'm working with the last stable release and haven;t checked out CVS yet.

--
Tristan 'Minty' Colgate
<[EMAIL PROTECTED]> | ICQ #154577755
---
  "I don't mean to sound bitter, cold, or cruel, but
 I am, so that's how it comes out"
- Bill Hicks


With some appropriate thought, and a basic understanding of how 
Asterisk handles call routing, this recent CVS note should point you 
in the right direction.

JT



From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-cvs] asterisk README.variables,1.9,1.10
Date: Wed, 12 Nov 2003 17:28:02 -0600 (CST)
Update of /usr/cvsroot/asterisk
In directory mongoose.digium.com:/tmp/cvs-serv6345
Modified Files:
README.variables
Log Message:
Improve documentation of ${SIPDOMAIN}
Index: README.variables
===
RCS file: /usr/cvsroot/asterisk/README.variables,v
retrieving revision 1.9
retrieving revision 1.10
diff -u -d -r1.9 -r1.10
--- README.variables11 Nov 2003 20:46:41 -  1.9
+++ README.variables12 Nov 2003 23:54:16 -  1.10
@@ -44,7 +44,7 @@
 ${DNID} Dialed Number Identifier
 ${RDNIS}Redirected Dial Number ID Service
 ${HANGUPCAUSE} Hangup cause on last PRI hangup
-${SIPDOMAIN}SIP domain (if appropriate)
+${SIPDOMAIN}SIP destination domain of an inbound call (if appropriate)
 There are two reference modes - reference by value and reference by name.
 To refer to a variable with its name (as an argument to a function that
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Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)

[Florian Overkamp]

At 09:52 18-11-2003 -0700, you wrote:
On Tue, 2003-11-18 at 08:53, Florian Overkamp wrote:
> Actually in this light it might be cute to have an 'uptime' counter inside
> asterisk (maybe a lastlog that can also show the reason of the last 
restart
> - was it a stop gracefully or did it just crash?) *grin*

Have you tried "show uptime" from the Asterisk CLI?
Ah very nice. I completely missed that :-)

Florian



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[Asterisk-Users] Bad DTMF detection

[Mark Farver]

We're still having problems with DTMF detection on our X100P cards. 

Incoming callers that hold down the "1" button for too long are being
connected to extension 11.  One would think fat fingers were uncommon,
but it happens to alot of people.

I suspected this was related to our having to increase the txgain, but I
tried turning it down with no effect.  I also tried disabling the
outgoing greeting, thinking it was interrupting the detector, with no
result.  Relax DTMF helps cellphone callers, but has no effect on the
double detect problem.

I cannot tell from the log before if asterisk is detecting a single "1"
and finding a nonexistent extension 1, or detecting the long keypress
as two keypresses (seems like two, with the "Oooh got something" message
being asterisk's way of saying at least on extension exists in the
context that begins with a 1).

DEBUG[14351]: File chan_zap.c, Line 1872 (zt_answer): Took Zap/27-1 off
hook
DEBUG[14351]: File channel.c, Line 952 (ast_settimeout): Scheduling
timer at 160 sample intervals
DEBUG[14351]: File chan_zap.c, Line 3388 (zt_read): DTMF digit: 1 on
Zap/27-1
DEBUG[14351]: File channel.c, Line 952 (ast_settimeout): Scheduling
timer at 0 sample intervals
DEBUG[14351]: File pbx.c, Line 1686 (ast_pbx_run): Oooh, got something
to jump out with ('1')!
DEBUG[14351]: File chan_zap.c, Line 3388 (zt_read): DTMF digit: 1 on
Zap/27-1
DEBUG[14351]: File chan_sip.c, Line 649 (create_addr): Setting NAT on
RTP to 0

--zapata.conf-- 
;rxgain=5%
txgain=7%
relaxdtmf=yes  
adsi=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhilebridged=yes
signalling=fxs_ks
callerid="Not sent"
context=default
group=2
channel=>25-27

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[Asterisk-Users] G723.1 Softphone for windows

[Hcqm]

Anybody knows about a FREE or GPL G723.1 capable softphone for windows?
thanks anyone.
Hector.
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[Asterisk-Users] "Unable to find path from G729A to ULAW" on Sipphone.com

[Steven Sokol]

I seem to be having a problem with transcoding and/or agreeing on a
valid codec.  I am running a new image pulled from CVS at 1:30 PM CST.
The issue occurs when I try to make a call to a toll-free number over
sipphone.com.

Here's what I see in the console:

NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1259545280]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from ULAW to G729A

Before somebody tells me "UTFG", I ALREADY HAVE.  Somebody else had a
similar issue last week and there was no real resolution posted.  So
here it is again.  I have all of the codecs that I support enabled in my
sip.conf.  Here is the relevant section:

;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
context = default   ; Default for incoming calls
srvlookup = yes ; Enable SRV lookups on outbound calls
pedantic = yes  ; Enable slow, pedantic checking for
Pingtel
;tos=lowdelay
;tos=184
maxexpirey=3600 ; Max length of incoming registration we allow
defaultexpirey=120  ; Default length of incoming/outoing
registration
;notifymimetype=text/plain  ; Allow overriding of mime type in
NOTIFY
;videosupport=yes   ; Turn on support for SIP video
disallow=all; Disallow all codecs
allow=ulaw  ; Allow codecs in order of preference
allow=alaw  ; Allow codecs in order of preference
allow=gsm
allow=ilbc

register => 17476692375:[EMAIL PROTECTED]/1101

[sipphone]
type=peer
username=17476692375
secret=[MYSECRET]
host=proxy01.sipphone.com
fromuser=SteveSokol
fromdomain=sipphone.com
canreinvite=no

; ==END OF SIP.CONF FILE===

The issue occurs whenever any calls that route over the sipphone peer
are made to a toll-free number.  The calling phone (either my GS100 or
my X-LITE softphone) rings two or three times then gives me busy.  Here
is the entire debug output:

-- Executing Dial("SIP/1101-1f83",
"SIP/[EMAIL PROTECTED]|20|tr") in new stack
-- Called [EMAIL PROTECTED]
NOTICE[1234379840]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1234379840]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from ULAW to G729A
-- SIP/sipphone.com-e7b3 is making progress passing it to
SIP/1101-1f83
-- SIP/sipphone.com-e7b3 answered SIP/1101-1f83
-- Attempting native bridge of SIP/1101-1f83 and
SIP/sipphone.com-e7b3
NOTICE[1242768320]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1242768320]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from ULAW to G729A
WARNING[1234379840]: File chan_sip.c, Line 1159 (sip_write): Asked to
transmit frame type 4, while native formats is 256 (read/write = 4/4)
  == Spawn extension (default, 918884510851, 1) exited non-zero on
'SIP/1101-1f83'

The problem does NOT occur when I call another sipphone.com user (i.e.
GS100 -> Asterisk -> Sipphone -> GS100).  Those calls go through just
fine.  The toll free calls were working last week.  Is it me, or is it
Sipphone.com?

Any suggestions would be greatly appreciated.

Steve


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RE: [Asterisk-Users] Asterisk GUI Client Released!!!

[mattf]

Hello,

The reason I used MySQL is for simplicity on the client end, there is a perl
module Net::MySQL that requires no extra libraries to be installed on the
machine, unlike Postgres module, and the Database routines used by this
program are hardly database intensive. It's easy enough to install the
Windows MySQL server in about 5 minutes on a machine somewhere on your
network to test this out if you like. If you don't use the optional call
loggin part of the client then you will probably never have more than 200
records in the MySQL database at any given time.

You can even run this on the same machine you are running PostgreSQL if you
like, the processor usage of the MySQL database machine if it is the only
thing running with 60 astGUIclient computers connected and 2 Asterisk
servers updating line records is about 0.2%, and the load average stays at
0.00 and that is on a PII 450. It would hardly hurt the performance of any
machine you put it on. I use PostgreSQL on the backend of our company's
financial transactional system (1 million+ transactions per week) and I love
it(can't wait to see if 7.4 is really any faster:) ), but PGSQL is overkill
for the client app, and the overhead on the client machine
side(libraries/modules or server app middleware) needed to facilitate
PostgreSQL is too much.

Thanks,

MATT---



-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 2:45 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk GUI Client Released!!!


mattf wrote:
> Hello,
> 
> I have finished my basic polishing of the Asterisk GUI client I have been
> writing in Perl/TK and have released a first beta version on sourceforge:
> 
> http://sourceforge.net/projects/astguiclient/
> 
> I am still working on a user manual for the application, but the code
works
> and we have been using the same basic client for the last month here at my
> company and it is working just fine.
> 
> I'm eager to hear what you all have to say about it, so send me those
> comments.
> 
> 

I wonder if there is hope for those of us who use Postgres and not MySQL?

I am anxious to run it, but probably not going to switch backend DB 
servers in order to do so.  I suspect there will be others with the same 
attitude.

Thx.

B.

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Re: [Asterisk-Users] manager.conf

[Ken Godee]

Steven Critchfield wrote:

On Tue, 2003-11-18 at 12:37, George Lin wrote:

Hi,

Do you know if we can use AGI or other script to handle the
asterisk events by using the existing asterisk manager process ?


AGI is for handling calls. AGI is to phone calls like CGI is to web page
requests.
There is a perl module to use in accessing manager events though. search
the archive for links to it. 


And also a very good python module available ..

http://sourceforge.net/projects/pyst/



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Re: [Asterisk-Users] Asterisk GUI Client Released!!!

[Areski]

Hi Matt,

After a first look to the screen shoot, it sounds incredible... 
Good work ;P

Will it work without Zaptel interface ?

Aresk

On Tue, 2003-11-18 at 20:03, mattf wrote:
> Hello,
> 
> I have finished my basic polishing of the Asterisk GUI client I have been
> writing in Perl/TK and have released a first beta version on sourceforge:
> 
> http://sourceforge.net/projects/astguiclient/
> 
> I am still working on a user manual for the application, but the code works
> and we have been using the same basic client for the last month here at my
> company and it is working just fine.
> 
> I'm eager to hear what you all have to say about it, so send me those
> comments.
> 
> 
> Here are the screen shots of the same application running on Linux and
> Windows:
> 
> http://www.freedomphones.net/astguiclient_linux.gif
> 
> http://www.freedomphones.net/astguiclient_windows.gif
> 
> MATT---
> 
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Re: [Asterisk-Users] Asterisk GUI Client Released!!!

[Steven Critchfield]

On Tue, 2003-11-18 at 13:44, Brian Capouch wrote:
> mattf wrote:
> > Hello,
> > 
> > I have finished my basic polishing of the Asterisk GUI client I have been
> > writing in Perl/TK and have released a first beta version on sourceforge:
> > 
> > http://sourceforge.net/projects/astguiclient/
> > 
> > I am still working on a user manual for the application, but the code works
> > and we have been using the same basic client for the last month here at my
> > company and it is working just fine.
> > 
> > I'm eager to hear what you all have to say about it, so send me those
> > comments.
> > 
> > 
> 
> I wonder if there is hope for those of us who use Postgres and not MySQL?
> 
> I am anxious to run it, but probably not going to switch backend DB 
> servers in order to do so.  I suspect there will be others with the same 
> attitude.

While I am of the same attitude, you need to look at the source. Since
it is written in perl/tk, all that should have to be changed is the
connect command, and maybe a few queries if they strayed very far into
the mysqlisms. This is a case again of how OSS and self sufficient users
are a good match. 

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Low Volume X100P

[Kevin]

I’ve tried that, all it seems to do
is distort the audio quality and add much more echo.  I see others have had this same problem and I
was wondering if it was resolved or just living with it.

 

-Original Message-
From: Josh J. Zuerner
[mailto:[EMAIL PROTECTED] 
Sent: Tuesday, November 18, 2003 9:17 AM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Low
Volume X100P

 



Yes, but, just increase the rxgain
in your zapata.conf and it will likely take care of the issue.





 





Josh







- Original Message - 





From: Kevin 





To: [EMAIL PROTECTED] 





Sent: Tuesday, November 18, 2003 9:12 AM





Subject:
[Asterisk-Users] Low Volume X100P





 





Has
anyone experienced low volume with the X100P FXO card?

Re: [Asterisk-Users] Radius on *

[Jan Janak]

On 17-11 16:33, Jeremy McNamara wrote:
> Sebastian Nocetti wrote:
> 
> >Does Asterisk support Radius accounting?
> > 
> >
> 
> No and there is absolutely no need for it to.   RADIUS is not anything 
> that should have ever been deployed in a VoIP environment.  

  You would be surprised how many people find RADIUS support useful in
  SIP Express Router and actually _use_ it. It is one of the most
  desired features, I am not kidding.

  Also IETF recently decided to standardize using of RADIUS with SIP,
  mainly because there is a huge user base.

   Jan.
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Re: [Asterisk-Users] G723.1 Softphone for windows

[Dan]

Hi,

- Original Message - 
From: "Hcqm" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 18, 2003 10:02 PM
Subject: [Asterisk-Users] G723.1 Softphone for windows


> Anybody knows about a FREE or GPL G723.1 capable softphone for windows?
> thanks anyone.
> Hector.

Netmeeting can use G.723 with H.323
I doubt that there is any other free/GPL softphone, because of the G.723
licensing mode

BR,
Dan

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RE: [Asterisk-Users] Asterisk GUI Client Released!!!

[mattf]

Hello,

Thanks :)

Sorry, it will not work out of the box without a zaptel device, you will
have to go in and tinker with the code a bit if you want it to work. Also,
some of the functions only work with Zap devices(call recording). Everything
I do with Asterisk is with SIP devices and T1s through Digium cards, so
haven't much use for a VOIP-only centric client app, but this code is
adaptable so you could definately change it to suit your needs. I'm know
there are several people out there that use asterisk without Zap devices so
there may be a strong enough desire for it. If anyone is interested in
modifying the code for non-Zap systems just let me know and I'll give you
all the help I can.

MATT---


-Original Message-
From: Areski [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 3:19 PM
To: Asterisk-Users Mailing-list
Subject: Re: [Asterisk-Users] Asterisk GUI Client Released!!!


Hi Matt,

After a first look to the screen shoot, it sounds incredible... 
Good work ;P

Will it work without Zaptel interface ?

Aresk

On Tue, 2003-11-18 at 20:03, mattf wrote:
> Hello,
> 
> I have finished my basic polishing of the Asterisk GUI client I have been
> writing in Perl/TK and have released a first beta version on sourceforge:
> 
> http://sourceforge.net/projects/astguiclient/
> 
> I am still working on a user manual for the application, but the code
works
> and we have been using the same basic client for the last month here at my
> company and it is working just fine.
> 
> I'm eager to hear what you all have to say about it, so send me those
> comments.
> 
> 
> Here are the screen shots of the same application running on Linux and
> Windows:
> 
> http://www.freedomphones.net/astguiclient_linux.gif
> 
> http://www.freedomphones.net/astguiclient_windows.gif
> 
> MATT---
> 
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Re: [Asterisk-Users] 3Com NBX phones

[Clif Jones]

We still have a few of the 3com phones in use at our company but we do 
not support them with our
SIP products.  The 3com phone was meant to be a PBX feature phone as you 
stated and as a result
the flash ROM and RAM was not beefy enough to support the SIP protocol 
as it matured.  The last
ROM update we got before they pulled the plug was still an old pre 3261 
bis standard which has problems
with voicemail and other centrex like features.  The sad part is that 
the phone HW had a good speakerphone
and booted faster than any other phone that we have tested (about 2 
seconds).

Steve Totaro wrote:

3com made a phone that supported SIP but was discontinued.  From what I have
heard, the hardware was exactly the same, just a different BIOS.  I would
like to try or hear of someone trying to flash an NBX phone into a SIP
phone.
Here is a link to one that sold on Ebay.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3057972164&category=11909
We use an NBX here in my office and the business phones are great.

No worry though, 3com's new product VCX7000 (
http://drs.yahoo.com/S=2766679/K=3com+vcx/v=2/SID=e/l=WS1/R=1/H=0/*-http://www.3comvcx.net/
 )
line is basically going to be an Asterisk system (plus a large fortune) and
will even run Linux (NBX runs VxWorks)  Work off of SIP and the phones
should be one step up from the current ones.  I am sure they will attempt to
make everything proprietary but going the direction of SIP is a very good
sign.
Thanks,
Steve Totaro
- Original Message - 
From: "Andrew Nelson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 17, 2003 5:02 PM
Subject: [Asterisk-Users] 3Com NBX phones

 

Has there been any luck getting the 3Com NBX series phones to work with
Asterisk?
Thanks!

-Andrew
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