Re: [Asterisk-Users] three way calling and cisco ata 186

[Thomas Dingermann]

Pavel Zheltouhov wrote:
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk  as pbx. I need feature called as 'three way calling' or 
'transfer with consultation'. Registering,calling and 'blind transfer' 
work fine.

Same here
Is this feature provided by sip clients or by asterisk itself ?
What I have to configure in ATA   and what keys I have to press
on my phones ?
Three way calling is enabled by setting  bits 28-29 to 10
 ( Select the Cisco VG248 Style for mid-call services.
 as described at 
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a00800c4d1f.html#42433 

But it seems not works,I always get conference call with 3 persons.

Same here - and if you hang up, the call is not transferred...
Something goes wrong when using the Hook-Flash with ATA/Asterisk.
Any solutions out there?

Thomas

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Re: [Asterisk-Users] Virtual fax on the Asterisk box

[Dan]

Hi Jim,

Thank you for your detailed answer.

> I doubt anything is available (yet) that speaks iax/iax2, but
> sip or h.323 should be supported.  Just make sure it can take
> a g.711 call and act like a fax machine; and take a bitmap
> and generate a g.711 call to send that document.
>
> If all you can find is a g.711 to t.38 solution, openh323 has
> support for t.38 to eg hylafax.  A bit of a rube-goldberg,
> but it ought to work.

I think to a SIP or IAX user agent ( I don't want to use H.323) with only
G.711 support and able to emulate a virtual modem who then be used by a
standard FAX application (Linux or Windows based)

>
> The biggest issue with these kinds of setups is latency.
> The fax protocols, as you can imagine, have latency
> requriements that are more stringent than voice calls.
I need it only on the LAN, with a PSTN connection through a X100P card, so
latency must not be an issue.

BR,
Dan


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[Asterisk-Users] msn

[Kelvin Chua]

hi guys,
 
have any of you guys managed to use msn 
messenger to authenticate with asterisk using its DNS name? based on my 
experience with other sip proxies, msn will not authenticate if it sees a 
different realm than the realm of the client. one workaround i did was to edit 
the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. 
after this, asterisk would send a 401 to the register message, at this point, 
i'm quite stuck. i also noticed that the nonce field 
is randdata? compared to iptel.org's ser and vovida's vocal. i notice 
a lot of difference on the sip messages composition. i'm running 0.4.0. 

Re: [Asterisk-Users] PCI Master Abort

[frank . barthe]

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Derek Beaumont
> Sent: Monday, July 07, 2003 4:15 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] PCI Master Abort
>
> I am always getting multiple PCI Master Abort messages when I try to
> plug in a second TDM400P.
> I have asked this question before, but I nothing really solved my
> problem and I just put it on the back burner for a while.
> I am at a point where this is a crucial issue.

I do have the same PCI Master abort message with a Wildcard S400P

It seems this is NOT an IRQ problem :
I did change the IRQ in the BIOS :
- manual assignation for all the PCI boards
- automatic assignation for the Wildcard
then verificate the status of the PCI devices :
- more /proc/pci
- just to list the IRQ really assigned, the memory I/O addresses, ...
All seems to be correct

Then launch the modules :
modprobe zaptel
modprobe wcfxo
modprobe wcfxs

But here the error message still appears !

I'm interested in your solution if you solve the pb !

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Re: [Asterisk-Users] Loaded latest CVS and get Broken PIPE!!!

[Mark Spencer]

If you type "ls /proc//fds" do you see an unusually
high (> 200 or so) file descriptors?  if so, what do they say?

Mark

On Mon, 7 Jul 2003, Alex Lopez wrote:

> I updated to the latest CVS about 4 hours ago.  If I let asterisk run,
> connect a few times via -r after about one hour the system does not
> let me in and it starts sucking up resources. One time it spawned about
> 100 processes. Had to do a reboot.
>
>
>
> This time it just did not let me in and started using up CPU time.
>
>
>
> Any ideas
>
>
>
>

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Re: [Asterisk-Users] Integratting * With Database(Newbie)

[God Knows Well]

Hi Steven

Thanx for your reply . U said i can store data in Postgres would u plz me 
tell the steps to Configure it ,as i am newbie i didnt understand properly 
ur reply if u want i can send u the whole scenario.

Thanx in Advance

Syed Obaid Amin

From: Steven Critchfield <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Integratting *  With Database(Newbie)
Date: 05 Jul 2003 11:13:35 -0500
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On Sat, 2003-07-05 at 08:42, God Knows Well wrote:
> Hi
>
> I think * supports database integration i would be thankful if anyone 
help
> me to configure my asterisk box with database support. One more think 
can i
> store Log in database.

asterisk supports a couple of things being integrated with databases,
but probably not what you are thinking about. Currently you can store
data and get to it via the PSQL application for integration with
postgres. Or you can use the Berkly DB code and the various DB commands.
CDR can be pushed out to a mysql database. From AGI you can connect to
any database you have linux code to support.
The config files are flat files and there is no integration with a
database to make them available that way.
--
Steven Critchfield <[EMAIL PROTECTED]>
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RE: [Asterisk-Users] ATA 186 in Australia

[Adam Goryachev]

Talk to Tony from Action Computers in Sydney (02) 9281 3988 and tell him I
sent you, or else LAN Systems (Denis Valente) or any Tech Pacific
reseller...

I bought one from Tony, but I don't remember which one I ended up with. The
easy way to tell is that one model is 'in stock' while the other was a 8
week lead time for me...

Regards,
Adam

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Steven Honson
> Sent: Tuesday, 8 July 2003 9:40 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] ATA 186 in Australia
>
>
> Hi All,
>
> I'm looking at setting up a Asterisk system, and hope to use ATA 186's
> with it.
> Im in Australia, and am getting mixed answers to if its the I1 or I2 i
> need, does anyone have any experience with using ATA 186's in Australia
> Also, can anyone recommend a good place to obtain these locally?
>
> Cheers,
> Steven
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[Asterisk-Users] Loaded latest CVS and get Broken PIPE!!!

[Alex Lopez]

I updated to the latest CVS about 4 hours ago.  If I let
asterisk run, connect a few times via –r after about one hour the
system does not let me in and it starts sucking up resources. One time it
spawned about 100 processes. Had to do a reboot.

 

This time it just did not let me in and started using up CPU
time.

 

Any ideas

 

[Asterisk-Users] Problems with Hangup Detection in VoiceMail2.

[Fred Ziegler]

Hi.
Has anyone experienced hangup detection problems with the VoiceMail2 app?
I have a console phone on the FXS port.  When I call a SIP phone, and get
its voicemail greeting, I can enter the VoiceMail2 app, leave a message,
and then hit # to stop message recording.

Recording does stop, but the channel stays up inside the VoiceMail2 app
(as shown by a "show channels" command) for about 60 seconds.  Then it
drops.

Is this expected behavior?  If not, how can I make it disconnect the
call sooner?

I get the very same behavior in all scenarios:
 a) if calling from a console phone to a SIP phone.
 b) if calling in on a POTS line, connected to an FXO card.

a portion of my extensions.conf file is shown below:
   .
   .
exten => 304,1,Dial(SIP/304,16) ;fred
exten => 304,2,Voicemail2(u304)
exten => 304,3,Hangup
   .
   .


--
Regards,
Fred
-_=o&o>_---
Fred R. Ziegler BMWOA  #77929   AMA #631103
<[EMAIL PROTECTED]> BMWRA  #22468   '97 R1100RTL
W. Medford, MA 02155YB #235  IBA #6190  '72 R60/5 swb
... holder of the Ultra-Prestigious MA IBMWR plate!! :-)

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[Asterisk-Users] asterisk and uclinux

[johncn]

Hello,every one! 
I would like to know if asterisk could run under 
uclinux.
Regards.

Re: [Asterisk-Users] Lot's of errors and warnings.

[Steven Critchfield]

Do you have the source that your kernel was compiled from? Do you at
least have the appropriate headers for you kernel and the config file
that was used?

On Mon, 2003-07-07 at 18:28, marrandy wrote:
> # make clean ; make install
> rm -f torisatool makefw tor2fw.h
> rm -f zttool
> rm -f *.o ztcfg tzdriver sethdlc
> rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
> rm -f gendigits tones.h
> rm -f libtonezone*
> rm -f tor2ee
> rm -f core
> cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -DSTANDALONE_ZAPATA   -c -o 
> gendigits.o gendigits.c
> cc -o gendigits gendigits.o -lm
> ./gendigits
> gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB 
> -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes 
> -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I 
> /usr/src/linux/include -I/usr/src/linux/include/net   -DECHO_CAN_MARK2 
> -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 
> -DSTANDALONE_ZAPATA -c zaptel.c
> In file included from /usr/include/linux/prefetch.h:13,
>  from /usr/include/linux/list.h:6,
>  from /usr/include/linux/module.h:11,
>  from zaptel.c:35:
> /usr/include/asm/processor.h:55: `CONFIG_X86_L1_CACHE_SHIFT' undeclared here 
> (not in a function)
> /usr/include/asm/processor.h:55: requested alignment is not a constant
> In file included from /usr/include/linux/module.h:20,
>  from zaptel.c:35:
> /usr/include/linux/modversions.h:1:2: #error 
> "==="
> /usr/include/linux/modversions.h:2:2: #error "You should not include 
> /usr/include/{linux,asm}/ header"
> /usr/include/linux/modversions.h:3:2: #error "files directly for the 
> compilation of kernel modules."
> /usr/include/linux/modversions.h:4:2: #error ""
> /usr/include/linux/modversions.h:5:2: #error "glibc now uses kernel header 
> files from a well-defined"
> /usr/include/linux/modversions.h:6:2: #error "working kernel version (as 
> recommended by Linus Torvalds)"
> /usr/include/linux/modversions.h:7:2: #error "These files are glibc internal 
> and may not match the"
> /usr/include/linux/modversions.h:8:2: #error "currently running kernel. They 
> should only be"
> /usr/include/linux/modversions.h:9:2: #error "included via other system header 
> files - user space"
> /usr/include/linux/modversions.h:10:2: #error "programs should not directly 
> include  or"
> /usr/include/linux/modversions.h:11:2: #error " as well."
> /usr/include/linux/modversions.h:12:2: #error ""
> /usr/include/linux/modversions.h:13:2: #error "To build kernel modules please 
> do the following:"
> /usr/include/linux/modversions.h:14:2: #error ""
> /usr/include/linux/modversions.h:15:2: #error " o Have the kernel sources 
> installed"
> /usr/include/linux/modversions.h:16:2: #error ""
> /usr/include/linux/modversions.h:17:2: #error " o Make sure that the symbolic 
> link"
> /usr/include/linux/modversions.h:18:2: #error "   /lib/modules/`uname 
> -r`/build exists and points to"
> /usr/include/linux/modversions.h:19:2: #error "   the matching kernel source 
> directory"
> /usr/include/linux/modversions.h:20:2: #error ""
> /usr/include/linux/modversions.h:21:2: #error " o Now copy 
> /boot/vmlinuz.version.h to"
> /usr/include/linux/modversions.h:22:2: #error "   /lib/modules/`uname 
> -r`/build/include/linux/version.h"
> /usr/include/linux/modversions.h:23:2: #error ""
> /usr/include/linux/modversions.h:24:2: #error " o When compiling, make sure to 
> use the following"
> /usr/include/linux/modversions.h:25:2: #error "   compiler option to use the 
> correct include files:"
> /usr/include/linux/modversions.h:26:2: #error ""
> /usr/include/linux/modversions.h:27:2: #error "   -I/lib/modules/`uname 
> -r`/build/include"
> /usr/include/linux/modversions.h:28:2: #error ""
> /usr/include/linux/modversions.h:29:2: #error "   instead of"
> /usr/include/linux/modversions.h:30:2: #error ""
> /usr/include/linux/modversions.h:31:2: #error "   -I/usr/include/linux"
> /usr/include/linux/modversions.h:32:2: #error ""
> /usr/include/linux/modversions.h:33:2: #error "   Please adjust the Makefile 
> accordingly."
> /usr/include/linux/modversions.h:34:2: #error 
> "==="
> In file included from /usr/include/linux/module.h:297,
>  from zaptel.c:35:
> /usr/include/linux/version.h:2:2: #error 
> "==="
> /usr/include/linux/version.h:3:2: #error "You should not include 
> /usr/include/{linux,asm}/ header"
> /usr/include/linux/version.h:4:2: #error "files directly for the compilation 
> of kernel modules."
> /usr/include/linux/version.h:5:2: #error ""
> /usr/include/linux/version.h:6:2: #error "glibc now uses kernel header files 
> from a well-defined"
> /usr/include/linux/version.h:7:2: #error "working kernel version (as 
> recommended by Linus Torvalds)"
> /usr/include/linux/version.h:8:2: #error "These files ar

Re: [Asterisk-Users] Dial plan doesn't seem to save properly

[Steven Critchfield]

Read the original config file, especially near the top. It isn't
implemented yet.

On Mon, 2003-07-07 at 17:23, [EMAIL PROTECTED] wrote:
> When I first to the "add extension" the "show dialplan" has the lines that
> say "SIP/"  but after I do a "save dialplan" and a "stop gracfully" and
> restart  the lines with "SIP/" are gone.
> 
> 
> 
> "Show dialplan" before:
> 
> 
> asterisk01*CLI>
> [ Context 'default' created by 'pbx_config' ]
>   '0001' => -1. Dial()
> [pbx_config]
> 1. Dial(SIP/mvickers)
> [pbx_config]
>   '0002' => -1. Dial()
> [pbx_config]
> 1. Dial(SIP/jgb)
> [pbx_config]
>   '0003' => -1. Dial()
> [pbx_config]
> 1. Dial(SIP/markw)
> [pbx_config]
>   '0004' => -1. Dial()
> [pbx_config]
> 1. Dial(SIP/rodr)
> [pbx_config]
> asterisk01*CLI>
>   Include =>'demo'
> [pbx_config]
> 
> [ Context 'parkedcalls' created by 'res_parking' ]
>   '701' =>  1. ParkedCall(701)
> [res_parking]
> 
> 
> *
> "Show dialplan" after:
> *
> 
> 
> [ Context 'default' created by 'pbx_config' ]
>   '0001' => -1. Dial()
> [pbx_config]
>   '0002' => -1. Dial()
> [pbx_config]
>   '0003' => -1. Dial()
> [pbx_config]
>   '0004' => -1. Dial()
> [pbx_config]
> 
>   Include =>'demo'
> [pbx_config]
> 
> [ Context 'parkedcalls' created by 'res_parking' ]
>   '701' =>  1. ParkedCall(701)
> [res_parking]
>   '702' =>  1. ParkedCall(702)
> [res_parking]
> 
> 
> 
> 
> Thanks in advance
> 
> Mark Vickers, RealNetworks Inc.  Desk: (206) 674-2391  Fax: (206)674-3588
> 
> 
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Re: [Asterisk-Users] PCI Master Abort

[Leo Ann Boon]

A general tip:
- use APIC even on a single processor system
APIC (Advanced Programmable Interrupt Controller?) or more specifically 
IO-APIC, is a feature on newer motherboards (>=2000) that allows the OS 
to allocate more than 15 IRQs. Windows XP depends on this feature to 
solve the IRQ sharing issue, APIC is a requirement for Windows logo. In 
Linux APIC is a little murky, it's only turned on if you're using an SMP 
kernel. I've found that even on a single processor system, using an SMP 
kernel will turn on IO-APIC. During kernel config, there're 2 options 
for uniprocess APIC and IO-APIC. For some reasons, I can't get the 
uniprocess IO-APIC to work. But if I set the SMP option, it does come 
on. On my nForce Athlon, using RH's SMP kernel will allow the OS to 
allocate up to IRQ 19 without any sharing. IIRC, the maximum is 
24/processor.

FYI.

Joe Antkowiak wrote:

You can force IRQs in your BIOS config, I would set each card to its own IRQ
that doesn't get shared with anything else.  Disable your serial and
parallel ports if you aren't using them and use 3,4,5,7
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek Beaumont
Sent: Monday, July 07, 2003 4:15 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PCI Master Abort
I am always getting multiple PCI Master Abort messages when I try to
plug in a second TDM400P.
I have asked this question before, but I nothing really solved my
problem and I just put it on the back burner for a while.
I am at a point where this is a crucial issue.
I have read that the Zaptel devices share an IRQ, is this causing the
problem?
Is there a way that I can manually change the IRQs of the devices?
By the way, my hardware situation is as follows:
2 X100P
2 TDM400P
Any help is always appreciated.

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[Asterisk-Users] Dial plan doesn't seem to save properly

[mvickers]

When I first to the "add extension" the "show dialplan" has the lines that
say "SIP/"  but after I do a "save dialplan" and a "stop gracfully" and
restart  the lines with "SIP/" are gone.



"Show dialplan" before:


asterisk01*CLI>
[ Context 'default' created by 'pbx_config' ]
  '0001' => -1. Dial()
[pbx_config]
1. Dial(SIP/mvickers)
[pbx_config]
  '0002' => -1. Dial()
[pbx_config]
1. Dial(SIP/jgb)
[pbx_config]
  '0003' => -1. Dial()
[pbx_config]
1. Dial(SIP/markw)
[pbx_config]
  '0004' => -1. Dial()
[pbx_config]
1. Dial(SIP/rodr)
[pbx_config]
asterisk01*CLI>
  Include =>'demo'
[pbx_config]

[ Context 'parkedcalls' created by 'res_parking' ]
  '701' =>  1. ParkedCall(701)
[res_parking]


*
"Show dialplan" after:
*


[ Context 'default' created by 'pbx_config' ]
  '0001' => -1. Dial()
[pbx_config]
  '0002' => -1. Dial()
[pbx_config]
  '0003' => -1. Dial()
[pbx_config]
  '0004' => -1. Dial()
[pbx_config]

  Include =>'demo'
[pbx_config]

[ Context 'parkedcalls' created by 'res_parking' ]
  '701' =>  1. ParkedCall(701)
[res_parking]
  '702' =>  1. ParkedCall(702)
[res_parking]




Thanks in advance

Mark Vickers, RealNetworks Inc.  Desk: (206) 674-2391  Fax: (206)674-3588


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[Asterisk-Users] ATA 186 in Australia

[Steven Honson]

Hi All,

I'm looking at setting up a Asterisk system, and hope to use ATA 186's 
with it.
Im in Australia, and am getting mixed answers to if its the I1 or I2 i 
need, does anyone have any experience with using ATA 186's in Australia
Also, can anyone recommend a good place to obtain these locally?

Cheers,
Steven
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[Asterisk-Users] SIP canreinvite=yes Broke?

[Dave Packham]

So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I 
cannot get the phones to talk/RTP to each other.  jtodd has had this problem in the 
past with the 186's.  Just wondering if anyone has a reason why "Cisco sometimes poop 
on reinvite"  is the Cisco code broke?  if so we can push on Cisco to fix it.  the U 
is a MAJOR Cisco shop so we have some puhs there.  if its * code,  I will offer up 
anything (within reason) to work this out.

This prob would be a major issue in rolling * out further if every call HAS to go thru 
the * server fro bridging.

Do other SIP hardsets have this problem?  sniff you calls to another SIP hardset and 
check ot see if RTP is coming from the * server of the other phone?

Thanks for any info I can get on this

Dave P


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[Asterisk-Users] Lot's of errors and warnings.

[marrandy]

# make clean ; make install
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -DSTANDALONE_ZAPATA   -c -o 
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB 
-I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes 
-fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I 
/usr/src/linux/include -I/usr/src/linux/include/net   -DECHO_CAN_MARK2 
-DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 
-DSTANDALONE_ZAPATA -c zaptel.c
In file included from /usr/include/linux/prefetch.h:13,
 from /usr/include/linux/list.h:6,
 from /usr/include/linux/module.h:11,
 from zaptel.c:35:
/usr/include/asm/processor.h:55: `CONFIG_X86_L1_CACHE_SHIFT' undeclared here 
(not in a function)
/usr/include/asm/processor.h:55: requested alignment is not a constant
In file included from /usr/include/linux/module.h:20,
 from zaptel.c:35:
/usr/include/linux/modversions.h:1:2: #error 
"==="
/usr/include/linux/modversions.h:2:2: #error "You should not include 
/usr/include/{linux,asm}/ header"
/usr/include/linux/modversions.h:3:2: #error "files directly for the 
compilation of kernel modules."
/usr/include/linux/modversions.h:4:2: #error ""
/usr/include/linux/modversions.h:5:2: #error "glibc now uses kernel header 
files from a well-defined"
/usr/include/linux/modversions.h:6:2: #error "working kernel version (as 
recommended by Linus Torvalds)"
/usr/include/linux/modversions.h:7:2: #error "These files are glibc internal 
and may not match the"
/usr/include/linux/modversions.h:8:2: #error "currently running kernel. They 
should only be"
/usr/include/linux/modversions.h:9:2: #error "included via other system header 
files - user space"
/usr/include/linux/modversions.h:10:2: #error "programs should not directly 
include  or"
/usr/include/linux/modversions.h:11:2: #error " as well."
/usr/include/linux/modversions.h:12:2: #error ""
/usr/include/linux/modversions.h:13:2: #error "To build kernel modules please 
do the following:"
/usr/include/linux/modversions.h:14:2: #error ""
/usr/include/linux/modversions.h:15:2: #error " o Have the kernel sources 
installed"
/usr/include/linux/modversions.h:16:2: #error ""
/usr/include/linux/modversions.h:17:2: #error " o Make sure that the symbolic 
link"
/usr/include/linux/modversions.h:18:2: #error "   /lib/modules/`uname 
-r`/build exists and points to"
/usr/include/linux/modversions.h:19:2: #error "   the matching kernel source 
directory"
/usr/include/linux/modversions.h:20:2: #error ""
/usr/include/linux/modversions.h:21:2: #error " o Now copy 
/boot/vmlinuz.version.h to"
/usr/include/linux/modversions.h:22:2: #error "   /lib/modules/`uname 
-r`/build/include/linux/version.h"
/usr/include/linux/modversions.h:23:2: #error ""
/usr/include/linux/modversions.h:24:2: #error " o When compiling, make sure to 
use the following"
/usr/include/linux/modversions.h:25:2: #error "   compiler option to use the 
correct include files:"
/usr/include/linux/modversions.h:26:2: #error ""
/usr/include/linux/modversions.h:27:2: #error "   -I/lib/modules/`uname 
-r`/build/include"
/usr/include/linux/modversions.h:28:2: #error ""
/usr/include/linux/modversions.h:29:2: #error "   instead of"
/usr/include/linux/modversions.h:30:2: #error ""
/usr/include/linux/modversions.h:31:2: #error "   -I/usr/include/linux"
/usr/include/linux/modversions.h:32:2: #error ""
/usr/include/linux/modversions.h:33:2: #error "   Please adjust the Makefile 
accordingly."
/usr/include/linux/modversions.h:34:2: #error 
"==="
In file included from /usr/include/linux/module.h:297,
 from zaptel.c:35:
/usr/include/linux/version.h:2:2: #error 
"==="
/usr/include/linux/version.h:3:2: #error "You should not include 
/usr/include/{linux,asm}/ header"
/usr/include/linux/version.h:4:2: #error "files directly for the compilation 
of kernel modules."
/usr/include/linux/version.h:5:2: #error ""
/usr/include/linux/version.h:6:2: #error "glibc now uses kernel header files 
from a well-defined"
/usr/include/linux/version.h:7:2: #error "working kernel version (as 
recommended by Linus Torvalds)"
/usr/include/linux/version.h:8:2: #error "These files are glibc internal and 
may not match the"
/usr/include/linux/version.h:9:2: #error "currently running kernel. They 
should only be"
/usr/include/linux/version.h:10:2: #error "included via other system header 
files - user space"
/usr/include/linux/version.h:11:2: #error "programs should not directly 
include  or"
/usr/include/linux/version.h:12:2: #error " as well."
/usr/include/linux/version

Re: [Asterisk-Users] overlap dialing on a pri span

[salmon]

Martin,

I probably should have mentioned that: overlapdial=yes was set in 
zapata.conf (I take it this option is inherited through all the
channels I configure in zapata.conf). I also did a fresh checkout today.

My guess is that the main problem for now lies in the fact that asterisk 
won't execute a dial application once it received the first digit. 
Apparently, the extension _X. won't spawn dial before asterisk hits 
the timeout:

exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,2 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,2  ; Set Response Timeout to 10 seconds
exten => _X.,1,Dial,Zap/g8/BYEXTENSION

I can see asterisk pick up:

-- Executing Answer("Zap/159-1", "") in new stack

the receive some digits

DEBUG[22551]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: 7 on Zap/159-1
DEBUG[22551]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: 8 on Zap/159-1
<...>

and seconds (!) later asterisk dials out

-- Executing Dial("Zap/159-1", "Zap/g8/BYEXTENSION") in new stack
-- Called g8/78997899
-- Channel 1, span 8 got hangup

Do you know why? Is there a minimum number of digits asterisk need for an
inital setup message?

Thilo

> overlapdial=yes in zapata.conf
> for those channels that you want the overlapdialing be activated.
> 
> By default only incoming overlap dialing is enabled.

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[Asterisk-Users] BudgeTone-100 Early Dial

[Stephen R. Besch]

Paul,

First, make sure that you use inband DTMF. As far as I know, out of band 
still does not work.  Second, make sure that the firmware is up to date. 
 The "silent DTMF" problem was fixed a few releases ago (at rev 
xx.xx.xx.60 I believe).

--
Stephen R. Besch, Ph.D.
SachsLab
320 Cary Hall
SUNY at Buffalo
Buffalo, NY 14214
(716) 829-3289 x106
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Re: [Asterisk-Users] Virtual fax on the Asterisk box

[James H. Cloos Jr.]

> "Dan" == Dan  <[EMAIL PROTECTED]> writes:

Dan> I have seen all of this on a Cisco 17xx router, including IVR
Dan> and sending faxes through e-mail, but it is far too expensive for
Dan> me...  Theoretically is possible to have let's say an IAX or SIP
Dan> software phone, even on a separate computer as Asterisk who can
Dan> handle faxes.

Yes, it is possible to make such a software only fax that can eg
convert between g.711 and either t.38 or something that software
like hylafax can connect to via a pty.

Some of the work has been done as free/open software, but it
would probably take as much more work to tie it together into
a single project.

The real problem is that the telcos that generated the standards
tied everything up with patents, so something that can be released
as source is hard to generate.

I have seen ad copy (found via google) suggesting the there are
solutions out there for what the OP wants to do.  It would not
surprise me at all were they as expensive as the cisco option,
just for software to run on a linux or bsd box.  But perhaps
someone does have something priced at a (near-)comodity level.

I doubt anything is available (yet) that speaks iax/iax2, but
sip or h.323 should be supported.  Just make sure it can take
a g.711 call and act like a fax machine; and take a bitmap
and generate a g.711 call to send that document.

If all you can find is a g.711 to t.38 solution, openh323 has
support for t.38 to eg hylafax.  A bit of a rube-goldberg,
but it ought to work.

The biggest issue with these kinds of setups is latency.
The fax protocols, as you can imagine, have latency
requriements that are more stringent than voice calls.

As a final note, modern fax machines have finally started
supporting v.34 and v.34bis fax.  HP's latest and imediately
previous lines of office jets are an example.  They also
support a 300 dpi colour mode that, if I understand correctly,
is no longer proprietory-per-vendor.  It would be useful to
have support for the higher dpi and faster modes in any g.711
software fax products.

-JimC


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Fw: [Asterisk-Users] IAX Bandwidth Question

[Jay Tyndall]

Subject: Re: [Asterisk-Users] IAX Bandwidth Question


 Hi,

 I have changed the codec to lpc10 (I see what they mean by Mr. Roboto!!)
and
 the ping times generally dont go over 300ms.
 It seems very odd that GSM saturates the link.

 I would love to give g.723.1 a try but have no idea where to get the
proprietary codec... can anyone help?

Thanks
 Jay

> - Original Message - 
> From: "WipeOut ." <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, July 07, 2003 10:14 PM
> Subject: Re: [Asterisk-Users] IAX Bandwidth Question
>
>
> > I have 2 asterisk systems connected by a 56kbps internet dialup (so at
> best 33.6k in both directions) using IAX and GSM...
> >
> > The one * box is at my home and the other is in the office (before
anyone
> freaks this is a test environment)... Provided nothing is using the line
at
> the same time I am able to carry 1 voice call over the dialup link with no
> real latency.. Ping times when no call is in session are about 165-175ms
and
> when a call is active 190-240ms.. I just ran a call for 5 mins to see if
the
> ping time climbed as you are experiencing and it didn't increase at all..
> >
> > So you may have some other issue causing the increase in ping times..
> >
> > Sorry I probably wasn't much help..
> >
> > > Hi,
> > >
> > > I am using IAX to communicate between 2 sites, each site is using a
> 256k/64k ADSL Connection.
> > >
> > >
> > > I have noticed that when I connect my ping time to my 1st hop jumps
from
> 30~ms to over 12,000ms in over a period of about 10 minutes, it just keeps
> climbing until the link is saturated.
> > >
> > > Naturally, there is a very long delay when speaking.
> > >
> > > What bandwidth would be adequate for IAX? or how can I tune my config
to
> work better with my current bandwidth situation.
> > > I am using GSM codec and bandwidth=low in iax.conf
> > >
> > > Thanks in advance.
> > >
> > Jay.
> > -- 
> > __
> > http://www.linuxmail.org/
> > Now with e-mail forwarding for only US$5.95/yr
> >
> > Powered by Outblaze
> > ___
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>

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[Asterisk-Users] conection with other PBX's

[Paulo Mannheimer]

I have a client which is willing to connect * with a siemens hicon PBX. 

 

Any experiences that you wish to share, including FX, ISDN
or E1 connectivity issues?

 

PauloHM

 

[Asterisk-Users] BudgeTone-100 Early Dial

[Paul Barrett]

Hi All

I have 3 GrandStream BudgeTone-100's which connect to an * box with a HFC-S
based ISDN card using ISDN4Linux.
I have setup the BudgeTone-100's to use Early Dial which for calling between
the three phones works well, but for the external calls using the following
extension
exten => _9.,1,Dial(Modem/g1:${EXTEN:1})
Only the first digit is dial on the ISDN Line.
Does anyone know of a way I can use the Early dial feature and still have
the external call dialout?

One other thing when I dial the extension for the Voicemail Admin or get
through to voice mail on a busy / unanswered line there is no sound produced
on the phone, any ideas?

Thanks in advance
Paul

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Re: [Asterisk-Users] Asterisk and VMWare

[Andy Powell]

Hi Dan,

For a totally unrelated reason I did this today. * runs fine here
under VMware, athough I haven't stressed it at all. 

Andy

*** REPLY SEPARATOR  ***

On 07/07/2003 at 19:07 Dan wrote:

>Hi,
>
>There is any experience using Asterisk with VMWare?
>I think about installing a virtual linux box over VMWare and then Asterisk
>over it.
>
>Thanks,
>Dan
>
>
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Re: [Asterisk-Users] Asterisk and VMWare

[Dan]

Hi Steven,

The PC is behind a NAT firewall/router, so this is not a real issue for the
moment.
It is online 24/7 for allmost 2 years now without any major problem.
It hosts my Home Automation engine and it must remain powered on full time.

I do not intend to install X just for VMWare.

Thanks,
Dan

- Original Message - 
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 11:12 PM
Subject: Re: [Asterisk-Users] Asterisk and VMWare


> On Mon, 2003-07-07 at 14:26, Dan wrote:
> > Hi,
> >
> > The reason I ask this is because I have a Win2K PC running 24/7 which
has
> > enough power left, but if I cannot use any of the Digium hardware from
> > inside VMWare then is useless.
>
> You can not have a win2k PC running 24/7 without being highly vulnerable
> to exploits. I won't even get into the even after patched it is still
> vulnerable argument. You would be better off running win2k under vmware,
> but then you would have to have X ontop of your phone system and that
> isn't good either.
>
> > - Original Message - 
> > From: "Gary Gapinski" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Monday, July 07, 2003 9:38 PM
> > Subject: Re: [Asterisk-Users] Asterisk and VMWare
> >
> >
> > > On Monday 07 July 2003 12:07, Dan wrote:
> > > > Hi,
> > > >
> > > > There is any experience using Asterisk with VMWare?
> > > > I think about installing a virtual linux box over VMWare and then
> > > > Asterisk over it.
> > >
> > > I use it with VMware, but Asterisk runs on the host OS, not in the VM.
> > >
> > > You should have no problem running Asterisk in a VM, but you would be
> > > unable to use any Digium or similar hardware, as VMware does not
expose
> > > such devices to a VM.
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> Steven Critchfield  <[EMAIL PROTECTED]>
>
> ___
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>
>


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Re: [Asterisk-Users] SIp Registration

[WipeOut .]

Not 100% sure here but its probably somthing to do with the fact that MS doesn't 
support MD5 and I think * makes use of md5 password hashing during authentication..

Maybe you can try adding auth=plaintext to that account in the sip.conf I know this 
option works in the iax.conf..

Later..

> I  use Windows Messenger ( I duck as to let the hurled penguins barely
> miss my head :-) ) and I am able to place and receive calls. So what is
> the problem you ask???  If I specify a password in the password field of
> WM I get a Proxy Authentication Error during SIP debug and I am not able
> to connect until I remove the secret=blah and do not specify a password.
> Has anyone had this problem before???
> 
>  
> 
>  
> 

-- 
__
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RE: [Asterisk-Users] PCI Master Abort

[Joe Antkowiak]

You can force IRQs in your BIOS config, I would set each card to its own IRQ
that doesn't get shared with anything else.  Disable your serial and
parallel ports if you aren't using them and use 3,4,5,7

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek Beaumont
Sent: Monday, July 07, 2003 4:15 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PCI Master Abort

I am always getting multiple PCI Master Abort messages when I try to
plug in a second TDM400P.
I have asked this question before, but I nothing really solved my
problem and I just put it on the back burner for a while.
I am at a point where this is a crucial issue.

I have read that the Zaptel devices share an IRQ, is this causing the
problem?
Is there a way that I can manually change the IRQs of the devices?

By the way, my hardware situation is as follows:
2 X100P
2 TDM400P

Any help is always appreciated.

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Re: [Asterisk-Users] PCI Master Abort

[Brancaleoni Matteo]

Zaptel hardware, rule #1
don't share irq, each card on it's own irq

(disable usb, sound, rtc and other fancy hw you don't need)

Matteo.

Il lun, 2003-07-07 alle 22:15, Derek Beaumont ha scritto:
> I am always getting multiple PCI Master Abort messages when I try to
> plug in a second TDM400P.
> I have asked this question before, but I nothing really solved my
> problem and I just put it on the back burner for a while.
> I am at a point where this is a crucial issue.
> 
> I have read that the Zaptel devices share an IRQ, is this causing the
> problem?
> Is there a way that I can manually change the IRQs of the devices?
> 
> By the way, my hardware situation is as follows:
> 2 X100P
> 2 TDM400P
> 
> Any help is always appreciated.
> 
> ___
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[Asterisk-Users] Asterisk crashing after Voicemail box creation

[BK [address only for mailing lists]]

Hi

I have just been struggling for four days to get SIP working and now as 
I created a voicemail box, Asterisk has become very unstable and it 
can't bridge SIP phone to SIP provider calls anymore.

Calling internally from one SIP phone to another works fine.

Calling internally from a SIP phone to an analog phone on a Zap channel 
and vice versa works fine.

Incoming PSTN calls delivered to a SIP phone also works fine.

Dialing out from an analog phone on a Zap channel using a SIP provider 
works fine as well.

HOWEVER,

when dialing out using a SIP provider (both Nikotel and iConnect) 
Asterisk cannot bridge the two legs of the call and all I get is silence.

here is what the console shows:

-- Executing Dial("SIP/Sip1-1862", "SIP/[EMAIL PROTECTED]|60|r") 
in new stack
-- Called [EMAIL PROTECTED]
-- SIP/nikotel-4815 is ringing
-- SIP/nikotel-4815 answered SIP/Sip1-1862
-- Attempting native bridge of SIP/Sip1-1862 and SIP/nikotel-4815
  == Spawn extension (internal, 00442071231234, 1) exited non-zero on 
'SIP/Sip1-1862'

NB: PSTN number edited

I have stop/started Asterisk and even rebooted the machine, but no 
change.

This problem has popped up after I created a voicemail box. When I 
tested the voicemail, at the moment when I tried to listen to the 
recording (VoicemailMain) Asterisk crashed. After restarting, I can now 
get into VoicemailMain without a crash, but there is now a problem with 
SIP and Asterisk crashes once in a while.

Any ideas?

thanks in advance
rgds
bk
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Re: [Asterisk-Users] Asterisk and VMWare

[Steven Critchfield]

On Mon, 2003-07-07 at 14:26, Dan wrote:
> Hi,
> 
> The reason I ask this is because I have a Win2K PC running 24/7 which has
> enough power left, but if I cannot use any of the Digium hardware from
> inside VMWare then is useless.

You can not have a win2k PC running 24/7 without being highly vulnerable
to exploits. I won't even get into the even after patched it is still
vulnerable argument. You would be better off running win2k under vmware,
but then you would have to have X ontop of your phone system and that
isn't good either. 

> - Original Message - 
> From: "Gary Gapinski" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, July 07, 2003 9:38 PM
> Subject: Re: [Asterisk-Users] Asterisk and VMWare
> 
> 
> > On Monday 07 July 2003 12:07, Dan wrote:
> > > Hi,
> > >
> > > There is any experience using Asterisk with VMWare?
> > > I think about installing a virtual linux box over VMWare and then
> > > Asterisk over it.
> >
> > I use it with VMware, but Asterisk runs on the host OS, not in the VM.
> >
> > You should have no problem running Asterisk in a VM, but you would be
> > unable to use any Digium or similar hardware, as VMware does not expose
> > such devices to a VM.
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> 
> 
> ___
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-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] PCI Master Abort

[Derek Beaumont]

I am always getting multiple PCI Master Abort messages when I try to
plug in a second TDM400P.
I have asked this question before, but I nothing really solved my
problem and I just put it on the back burner for a while.
I am at a point where this is a crucial issue.

I have read that the Zaptel devices share an IRQ, is this causing the
problem?
Is there a way that I can manually change the IRQs of the devices?

By the way, my hardware situation is as follows:
2 X100P
2 TDM400P

Any help is always appreciated.

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[Asterisk-Users] Follow-up -- Using Asterisk with Nikotel

[BK [address only for mailing lists]]

Hi

thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service.

I have drafted a mini-how-to which is available at

http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf

This is a first draft, I will amend this further, in particular the "verify and debug" section which is blank right now.


here is a plain text summary:

in sip.conf ...

--
; SIP Registration with Nikotel
;
register => myusername:[EMAIL PROTECTED]

; SIP peer definition
;
[nikotel]
type=friend
secret=mypassword
auth=md5
username=myusername
fromuser=myusername	; IMPORTANT! Nikotel requires this!
host=calamar0.nikotel.com
--

in extensions.conf

--
; Extensions
;
[globals]
CC => 81	; Local country code (here Japan)
PSTN => Zap/1	; incoming NTT line
REDPHONE => Zap/2	; internal analog line
FAXPHONE => Zap/3	; internal fax line
IPPHONE1 => SIP/Sip1	; IP phone #1
IPPHONE2 => SIP/Sip2	; IP phone #2

; International long distance through VoIP service
;
[voipintl]
exten => _00N.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r)
exten => _00N.,2,Congestion

[voipnatl]
;
; National long distance and local throuh VoIP service
;
exten => _0N.,1,Dial(SIP/${CC}${EXTEN:[EMAIL PROTECTED],60,r)
exten => _0N.,2,Congestion

[international]
include => voipintl

[national]
include => voipnatl

[incoming]
exten => s,1,Dial(${IPPHONE1}&${IPPHONE2),20,r)

[internal]
exten => 1001,1,Dial(${IPPHONE1},20,tr)
exten => 1001,2,Congestion
exten => 1002,1,Dial(${IPPHONE2},20,tr)
exten => 1002,2,Congestion
include => national
include => international
--

regards
bk

Re: [Asterisk-Users] Direct entry to your own voice mailbox

[The Traveller]

Hey Dan,

On Mon, Jul 07, 2003 at 20:42:16 +0300, Dan wrote:

> > > It is possible to have a time stamp in the recorded message? I want to
> know
> > > when the  message has been recorded.
> >
> > I think someone here was working on a patch for that, which was waiting
> > for prompts to be recorded.  Not sure of the current status.
> Why other prompts?
> There is an application available (DateTime) to say current date and time.
> It cannot be integrated in the Voicemail appication?

I guess it could.  It's just that the bigger patch this enhancement was
part of, required some new prompts for an extra menu one could enter
while playing a message.  If I remember correctly, Mark was thinking
about adding this functionality, although it was some time a go and
he seems very busy at the moment.



Grtz,

   Oliver
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[Asterisk-Users] SIp Registration

[Alex Lopez]

I  use Windows Messenger ( I duck as to let the hurled
penguins barely miss my head J ) and I am able to
place and receive calls. So what is the problem you ask???  If I specify a
password in the password field of WM I get a Proxy Authentication Error during
SIP debug and I am not able to connect until I remove the secret=blah and do
not specify a password.  Has anyone had this problem before???

 

 

Re: [Asterisk-Users] Asterisk and VMWare

[Gary Gapinski]

On Monday 07 July 2003 15:26, Dan wrote:
> The reason I ask this is because I have a Win2K PC running 24/7 which
> has enough power left, but if I cannot use any of the Digium hardware
> from inside VMWare then is useless.

If you have not yet purchased the VMware license, run Linux, Asterisk, 
VMware _for Linux_, and W2K within VMware (with the caveat that special 
hardware will likely not be supported within VMs).

Even if you have already purchased VMware for Microsoft Windows (the MS 
and Linux licenses are not interchangeable), contact VMware regarding a 
possible exchange.

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Re: [Asterisk-Users] Direct entry to your own voice mailbox

[Dan]

Thanks Florian,
Dan

- Original Message - 
From: "Florian Overkamp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 9:15 PM
Subject: Re: [Asterisk-Users] Direct entry to your own voice mailbox


> Citeren Dan <[EMAIL PROTECTED]>:
>
> > There is any possibility to dial a specific extension and then enter in
your
> > own mailbox (the one defined for that specific SIP phone) without asking
for
> > the exxtension number but only for the password?
> >
> > I want to be the same extension for all phones, not a specific one for
each
> > of them.
>
> Hi,
>
> this is one tool I have used a lot up to now; I have a simple conversion
> table: callerid -> voicemailbox. You could simply used ${CALLERID} or
> something, but I also have some remote cellphones and the likes...
>
> #!/usr/bin/php4 -q
> 
> // Set some parameters
> ob_implicit_flush(true);// Don't buffer output
> set_time_limit(0);  // This program may take forever
>
> // Setup file handles
> $stdin = fopen("php://stdin", "r");
> set_file_buffer($stdin, 0);
> $stderr = fopen("php://stderr", "a");
> set_file_buffer($stderr, 0);
>
> // Do function definitions before we start the main loop
> function read() {
> global $stdin;
> $input=fgets($stdin, 255);
> return str_replace("\n", "", $input);
> return $input;
> }
>
> function errlog($line) {
> global $stderr;
> fputs($stderr, $line."\n");
> }
>
> // parse agi headers into array $agi["callerid"]
> while ($env=read()) {
>   errlog($env);
>   $s = split(": ",$env);
>   $agi[str_replace("agi_","",$s[0])] = trim($s[1]);
>
>   echo "VERBOSE \"".$s[0].": ".$s[1]."\" 1\n";
>
>   if (($env == "") || ($env == "\n")) {
> break;
>   }
> }
>
>
> // Main run
> $clid = $agi["callerid"];
> switch($clid) {
>   // enter the mailbox number for each valid callerid
>   // prepend 's' if you wish to trust the callerid and skip the password
check
>   case "0612345678":$parms = "s1000"; break;
>   default: $parms = "0"; break;
> }
> if($parms != "") $parms = " $parms";
>
> echo "EXEC VoiceMailMain$parms\n";
>
> // Close file handles
> fclose($stdin);
> fclose($stderr);
> ?>
>
> -- 
> Met vriendelijke groet,
> Florian Overkamp
> ObSimRef BV
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>
>


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Re: [Asterisk-Users] Getting Started with Digium T100/E100 cards

[Stefano Finetti]

To enable any of TxxxP or ExxxP card you must compile the zaptel package
from cvs.

Then to enable the T100P or E100P you should load the wct1xxp module.

Then you can use the zttool (/sbin/zttool) to check the status of the cards
and of the spans you've configured in /etc/zaptel.conf

--
Stefano

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Re: [Asterisk-Users] System command..

[Tilghman Lesher]

On Monday 07 July 2003 13:23, Martin Pycko wrote:
> The system at the moment can run some program/script but there is no
> way to retrieve the results. Although you could have tried with
> ${ENV(VARIABLE)}

That won't work, either.  A program may change its own environmental
variables, but it is not possible to change the environment of the
parent process.

-Tilghman

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Re: [Asterisk-Users] Can't access outside voicemail services throughasterisk

[Martin Pycko]

Come on,

exten => *98,1,Dial,Zap/g1/BYEXTENSION
should work since it's old sytax.

It's more propable that you have that *98 in a diffrent context
than assigned for those channels or you don't have the group 1 defined
properly 

Martin

On 7 Jul 2003, Steven Critchfield wrote:

> On Mon, 2003-07-07 at 13:31, Derek Beaumont wrote:
> > I want to be able to check my Bell voicemail
> > (*98) using a phone attached to my asterisk box.
> >
> > In extensions.conf I have defined
> > exten=>*98,1,Dial,Zap/g1/BYEXTENSION
> >
> > However, when I dial *98, I just get a fast busy
> > signal.
> >
> > Is the * digit reserved for internal purposes?
>
> Not really, but likely you do not have a extension to deal with that
> option. Try...
> exten => *98,1,Dial(Zap/${TRUNK}/${EXTEN}
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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>

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Re: [Asterisk-Users] overlap dialing on a pri span

[Martin Pycko]

overlapdial=yes in zapata.conf
for those channels that you want the overlapdialing be activated.

By default only incoming overlap dialing is enabled.

regards
Martin

On 7 Jul 2003, Thilo Salmon wrote:

> Hi,
>
> I am lost trying to figure out how to enable overlap dialing for calls
> coming in and coing out on a pri span. DISA looked promising at first,
> but does not seem to support overlap dialing. Just picking up a call by
> and trying to dial out does not seem the way to do it either. I tried:
>
> [dialincontext]
> exten => 12341234,1,Goto(dialoutcontext,s,1)
>
> [dialoutcontext]
> exten => s,1,Wait,1 ; Wait a second, just for fun
> exten => s,2,Answer ; Answer the line
> exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
> exten => _X.,1,Dial,Zap/g8/BYEXTENSION
> exten => t,1,Dial,Zap/g8/BYEXTENSION
>
> in extensions.conf, hoping this would make asterisk dial out after
> reading a single digit. I can see asterisk detecting DTMF digits, but it
> won't dial out before the timeout kicks in and then hangs up right away.
>
> In the end I want to serve incoming callers with a dialtone, listen for
> DTMF digits and dial each digit seperately by submitting q.931
> information messages. According to Mark and the parts of libpri.h and
> app_dial.c I understood asterisk supports this way of "overlap dialing".
> Any ideas how to achieve this? A simple extensions.conf configured to
> handle overlap dialing could help me a lot.
>
> Thilo
>
>
>
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Re: [Asterisk-Users] Asterisk and VMWare

[Dan]

Hi,

The reason I ask this is because I have a Win2K PC running 24/7 which has
enough power left, but if I cannot use any of the Digium hardware from
inside VMWare then is useless.

Thank you,
Dan

- Original Message - 
From: "Gary Gapinski" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 9:38 PM
Subject: Re: [Asterisk-Users] Asterisk and VMWare


> On Monday 07 July 2003 12:07, Dan wrote:
> > Hi,
> >
> > There is any experience using Asterisk with VMWare?
> > I think about installing a virtual linux box over VMWare and then
> > Asterisk over it.
>
> I use it with VMware, but Asterisk runs on the host OS, not in the VM.
>
> You should have no problem running Asterisk in a VM, but you would be
> unable to use any Digium or similar hardware, as VMware does not expose
> such devices to a VM.
>
>
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>


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RE: [Asterisk-Users] System command..

[Wade Weppler]

It could be possible, but it might be simpler to try AGI and Asterisk-PERL.
It works great!

-wade

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of WipeOut .
> Sent: Monday, July 07, 2003 1:44 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] System command..
> 
> Can the system command be used to retrieve a variable from a mysql
> database using the mysql command line client??
> or would it be simpler to write some sort of AGI type application??
> --
> __
> http://www.linuxmail.org/
> Now with e-mail forwarding for only US$5.95/yr
> 
> Powered by Outblaze
> ___
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Re: [Asterisk-Users] Need a recommendation on a goodmotherboard/processor combination

[Steven Critchfield]

On Mon, 2003-07-07 at 13:29, Clay Graner wrote:
> I need a recommendation on a good motherboard/processor combination. 
> I would like a motherboard that has lots of PCI slots and works well
> with Asterisk without problems getting drivers working, etc. Onboard
> LAN would be nice to keep from using a slot.  Plan to use RedHat 8 for
> the OS.  
> 
>  
> 
> Please provide your input as soon as possible.  I need to build a demo
> box with 2 ports FXO and use the new 4 Port FXS card as well as
> connect a SIP phone.  

Boys and girls, this is why microsoft email should not be used. This
email is an exact duplicate of the one sent on Thursday. 

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Can't access outside voicemail servicesthrough asterisk

[Steven Critchfield]

On Mon, 2003-07-07 at 13:31, Derek Beaumont wrote:
> I want to be able to check my Bell voicemail 
> (*98) using a phone attached to my asterisk box.
> 
> In extensions.conf I have defined 
> exten=>*98,1,Dial,Zap/g1/BYEXTENSION
> 
> However, when I dial *98, I just get a fast busy 
> signal.
> 
> Is the * digit reserved for internal purposes?

Not really, but likely you do not have a extension to deal with that
option. Try...
exten => *98,1,Dial(Zap/${TRUNK}/${EXTEN}
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] overlap dialing on a pri span

[Thilo Salmon]

Hi, 

I am lost trying to figure out how to enable overlap dialing for calls
coming in and coing out on a pri span. DISA looked promising at first,
but does not seem to support overlap dialing. Just picking up a call by
and trying to dial out does not seem the way to do it either. I tried:

[dialincontext]
exten => 12341234,1,Goto(dialoutcontext,s,1) 

[dialoutcontext]
exten => s,1,Wait,1 ; Wait a second, just for fun 
exten => s,2,Answer ; Answer the line 
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => _X.,1,Dial,Zap/g8/BYEXTENSION
exten => t,1,Dial,Zap/g8/BYEXTENSION

in extensions.conf, hoping this would make asterisk dial out after
reading a single digit. I can see asterisk detecting DTMF digits, but it
won't dial out before the timeout kicks in and then hangs up right away.

In the end I want to serve incoming callers with a dialtone, listen for
DTMF digits and dial each digit seperately by submitting q.931
information messages. According to Mark and the parts of libpri.h and
app_dial.c I understood asterisk supports this way of "overlap dialing".
Any ideas how to achieve this? A simple extensions.conf configured to
handle overlap dialing could help me a lot.

Thilo



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[Asterisk-Users] One-way talk paths (without INVITE?) and other issues

[Moshe Yudkowsky]

I'm experiencing one-way voice paths, followed by a hangup on one 
softphoine and not the other. Also, caller ID has odd outputs -- and I 
wonder if the problems are related.

My configuration has Asterisk and a Linphone softphone running on the 
same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect 
to the Linphone instance.

When I call from the PC to Linphone:

* I call from the PC (user m12) to Linphone (usr m4), which rings

* I answer on Linphone

* Asterisk attempts to set up a talk path. Here's the output from 
Asterisk, with Linphone (m4) connecting to the PC (m12):

-- Called m4
-- SIP/m4-8f2b is ringing
-- Registered SIP 'm12' at 172.28.54.34 port 5060 expires 500
-- SIP/m4-8f2b answered SIP/m12-195f
-- Attempting native bridge of SIP/m12-195f and SIP/m4-8f2b


I don't know how the "registered" statement appreared in the middle.

At this point I can talk into the PC and hear it on Linphone -- but I 
cannot speak into Linphone and hear myself on the PC.  After about 10 
seconds, possibly less:

* The PC phone gets a hangup message (BYE).

* Linphone does *not* get a hangup message and remains offhook. Any 
attempt to call Linphone from the PC results in Asterisk routing the 
call to voicemail. I must manually hang up Linphone.

Oddly enough, the caller ID displayed by Linphone, and apparently sent 
by Asterisk, is incorrect. Instead of showing "m12" as the caller ID, 
Linphone receives "m1" as the caller ID:


INVITE sip:[EMAIL PROTECTED]:5062 SIP/2.0
Via: SIP/2.0/UDP x1.x2.x3.x4:5060;branch=z9hG4bK0c4d7e4c
From: "m1" ;tag=as62b91e33
To: ;tag=4210403538
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 159
(x1.x2.x3.x4 substituted for the actual IP address.)

To my fairly untrained eye, this looks like a legitimate proxy message 
but the caller ID is wrong. My SIP configuration file does contain both 
m1 and m12 as legitimate callers:

> [m1]
> type=friend
> username=m1
> host=dynamic
> permit=x1.x2.x3.0/16
> [m12]
> type=friend
> username=m1
> host=dynamic
> permit=x1.x2.x3.0/16
I also note the following Asterisk warnings. I cannot tell if they 
happen just before or just after I lose the one-way voice path:

WARNING[81926]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on 
call [EMAIL PROTECTED] for seqno 103 (Request)
WARNING[81926]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on 
call [EMAIL PROTECTED] for seqno 104 (Request)
Furthermore, I have yet to see a SIP channel disappear after a call is 
over. They are always listed as active, even hours later. Here are is 
the result of "show sip channels":

Peer User/ANRCall ID  Seq (Tx/Rx)  Lag  Jitter  Format
172.28.54.160m4  478c64565be  00104/0  0ms  ms  0
172.28.54.160m4  4d330ced01e  00104/1  0ms  ms  0
172.28.54.160m4  0bb7855f15b  00104/1  0ms  ms  0
172.28.54.160m4  3db8538b4b4  00104/1  0ms  ms  0
4 active SIP channel(s)


but none of these calls are "active" in any sense of the term that I can 
think of. I have tried to use the "sip show channel" command for further 
testing, but apparently I don't understand the syntax of the command -- 
"sip show channel 478c64565be" and "sip show channel m4/478c64565be" and 
"sip show channel SIP/m4-8f2b" all give the same error message, "no such 
SIP call ID 'whatever'". Either I don't understand the "sip show 
channels" command, or there's a bug.

My questions are:

* How is it that I get one-way voice paths? Is this a configuration 
problem? Are the INVITES not getting through but the voice paths 
established anyway?

* What's the problem with the incorrect caller IDs? I have *no* caller 
ID settings that I'm aware of in my *.conf files. The PC's program 
registers as "m12" but Asterisk sends "m1" as the name. PC-side 
debugging shows that the PC sends "[EMAIL PROTECTED]" as its name.

Although this feels like a bug, I strongly suspect that I'm missing some 
simple SIP configuration issue, but I haven't been able to track it down 
just yet.  And I'd like to clarify any other issues before starting on a 
bug hunt.

--
 Moshe Yudkowsky * http://www.Disaggregate.com
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Re: [Asterisk-Users] System command..

[Steven Critchfield]

On Mon, 2003-07-07 at 13:23, Martin Pycko wrote:
> The system at the moment can run some program/script but there is no way
> to retrieve the results. Although you could have tried with
> ${ENV(VARIABLE)}

This shouldn't work as the system command will run in it's own
environment and should not modify the parent process.

> On Mon, 7 Jul 2003, WipeOut . wrote:
> 
> > Can the system command be used to retrieve a variable from a mysql database using 
> > the mysql command line client??
> > or would it be simpler to write some sort of AGI type application??
> > --
> > __
> > http://www.linuxmail.org/
> > Now with e-mail forwarding for only US$5.95/yr
> >
> > Powered by Outblaze
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Asterisk and VMWare

[Gary Gapinski]

On Monday 07 July 2003 12:07, Dan wrote:
> Hi,
>
> There is any experience using Asterisk with VMWare?
> I think about installing a virtual linux box over VMWare and then
> Asterisk over it.

I use it with VMware, but Asterisk runs on the host OS, not in the VM.

You should have no problem running Asterisk in a VM, but you would be 
unable to use any Digium or similar hardware, as VMware does not expose 
such devices to a VM.


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[Asterisk-Users] Getting Started with Digium T100/E100 cards

[Langley, Sean]

Just purchased a couple of T100 and E100 cards in order to interface from
our company's proprietary system through a linux gateway.  I am new to
Asterisk and Digium.

After installing the T100 card, I went looking for drivers for this card.
Are the drivers built into the Asterisk application?  If so, how do I enable
the card through Asterisk?

Are there any good getting started documents for a newbie like myself?  I
have downloaded and read the Asterisk handbook, while helpful, it doesn't
answer all my questions.  (of course I could just be daft!)

Regards,

Sean Langley, P.Eng
Firmware Engineer
General Dynamics Canada
(403)730-1482
[EMAIL PROTECTED]

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[Asterisk-Users] Need a recommendation on a good motherboard/processor combination

[Clay Graner]

I need a recommendation on a good motherboard/processor
combination.  I would like a
motherboard that has lots of PCI slots and works well with Asterisk without
problems getting drivers working, etc. 
Onboard LAN would be nice to keep from using a slot.  Plan to use RedHat 8 for the OS.  

 

Please provide your input as soon as possible.  I need to build a demo box with 2 ports
FXO and use the new 4 Port FXS card as well as connect a SIP phone.  

 

Thanks,

Clay Graner

 

[Asterisk-Users] Can't access outside voicemail services through asterisk

[Derek Beaumont]

I want to be able to check my Bell voicemail 
(*98) using a phone attached to my asterisk box.

In extensions.conf I have defined 
exten=>*98,1,Dial,Zap/g1/BYEXTENSION

However, when I dial *98, I just get a fast busy 
signal.

Is the * digit reserved for internal purposes?

Any help is appreciated.

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Re: [Asterisk-Users] Network design question

[John Todd]

Hello!

My business is a wireless ISP.  I would like to offer voice to 
several business customers.  I have a * server, but still need some 
hardware cards for it.
I would like to provide individualized billing to these customers 
for their usage.  Most of these customers have Nortel PBX's.   Could 
I use a Cisco ATA  or a similar manufacturer to provide 1-2 line 
service?  I would like to get this set up correctly the first time. 
What would you suggest or has someone already done this that could 
explain better?

Thanks,

Bill
The ATA-186 works well, and it's price point is hard to beat.  I have 
several clients who use them with no difficulties.  If you have them 
behind NAT, however, they get a little fussy and you need to have 
them auto-reboot every once in a while.

JT
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Re: [Asterisk-Users] System command..

[Martin Pycko]

The system at the moment can run some program/script but there is no way
to retrieve the results. Although you could have tried with
${ENV(VARIABLE)}

Martin

On Mon, 7 Jul 2003, WipeOut . wrote:

> Can the system command be used to retrieve a variable from a mysql database using 
> the mysql command line client??
> or would it be simpler to write some sort of AGI type application??
> --
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Re: [Asterisk-Users] Direct entry to your own voice mailbox

[Florian Overkamp]

Citeren Dan <[EMAIL PROTECTED]>:

> There is any possibility to dial a specific extension and then enter in your
> own mailbox (the one defined for that specific SIP phone) without asking for
> the exxtension number but only for the password?
> 
> I want to be the same extension for all phones, not a specific one for each
> of them.

Hi,

this is one tool I have used a lot up to now; I have a simple conversion 
table: callerid -> voicemailbox. You could simply used ${CALLERID} or 
something, but I also have some remote cellphones and the likes...

#!/usr/bin/php4 -q


-- 
Met vriendelijke groet,
Florian Overkamp
ObSimRef BV
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Re: [Asterisk-Users] System command..

[Steven Critchfield]

On Mon, 2003-07-07 at 12:43, WipeOut . wrote:
> Can the system command be used to retrieve a variable from a mysql database using 
> the mysql command line client??
> or would it be simpler to write some sort of AGI type application??

  -= Info about application 'System' =- 

[Synopsis]:
  Execute a system command

[Description]:
  System(command): Executes a command  by  using  system(). Returns -1 on
failure to execute the specified command. If  the command itself executes
but is in error, and if there exists a priority n + 101, where 'n' is the
priority of the current instance, then  the  channel  will  be  setup  to
continue at that priority level.  Otherwise, System returns 0.


>From this it looks like there is no direct route to a asterisk variable.
Your only option would be a numeric return and then coding for each
n+101 priority. Better and easier to go AGI or postgres.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Network design question

[Steven Critchfield]

On Mon, 2003-07-07 at 12:33, Asterisk wrote:
> Hello!
> 
> My business is a wireless ISP.  I would like to offer voice to several 
> business customers.  I have a * server, but still need some hardware cards 
> for it.
> I would like to provide individualized billing to these customers for their 
> usage.  Most of these customers have Nortel PBX's.   Could I use a Cisco 
> ATA  or a similar manufacturer to provide 1-2 line service?  I would like 
> to get this set up correctly the first time.  What would you suggest or has 
> someone already done this that could explain better?

Since an ata can only do compression on one phone line at a time,
consider it to be nothing more than a single line device if bandwidth is
of concern. Also notice the various problems with SIP that crop up from
time to time. It is actually more useful to use IAX and a small PC with
TDM cards than an ATA.

As for the billing, you can track individual users of IAX in the CDR and
can easily use it this way for your billing. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Network design question

[Michael Kane]

Couple of things that come to mind.

What's the latency or your point to point wireless connections.
What type of voice interface do the PBX's have(analog, digital)?
If you plan on using a ATA like device and the Nortel phonesets are digital
you probably will run into echo problems.
What do you need the voice cards in the Asterisks for?  Interconnection to
the PSTN, to your switch?

Mike

- Original Message -
From: "Asterisk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 1:33 PM
Subject: [Asterisk-Users] Network design question


>
> Hello!
>
> My business is a wireless ISP.  I would like to offer voice to several
> business customers.  I have a * server, but still need some hardware cards
> for it.
> I would like to provide individualized billing to these customers for
their
> usage.  Most of these customers have Nortel PBX's.   Could I use a Cisco
> ATA  or a similar manufacturer to provide 1-2 line service?  I would like
> to get this set up correctly the first time.  What would you suggest or
has
> someone already done this that could explain better?
>
> Thanks,
>
> Bill
>
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Re: [Asterisk-Users] Asterisk and VMWare

[Steven Critchfield]

On Mon, 2003-07-07 at 11:48, WipeOut . wrote:
> If you want to use it for IP only it __MAY__ work but probably not all
> that well unless you have a really strong processor..
> 
> VMWare abstracts the hardware layer you will not be able to get it to
> communicate fully with any cards in order to connect phone lines..

You forgot to mention that as an application running in userland it is
possible to stall and hang while kernel or other less nice applications
hog the system. I use vmware on a regular basis and have seen times
where system speed inside of vmware is dependent on other factors like
mouse movement to force vmware to receive a few more time slices. 

> > Hi,
> > 
> > There is any experience using Asterisk with VMWare?
> > I think about installing a virtual linux box over VMWare and then Asterisk
> > over it.
> > 
> > Thanks,
> > Dan
> > 
> > 
> > ___
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-- 
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Re: [Asterisk-Users] Newbie Doubts

[Michael Kane]

Conceptually, one could purchase a Cisco(or Cisco like device) that has a
DVM module(digital voice module) and route the TDM traffic from the DVM out
the Ethernet port(VoIP).  In a previous life some could and did call this
device a channel bank.  Although I disagree.  Wade is right on the money.  A
traditional channel bank (T1, M13) will mux many to one from either
digital/analog or digital/ditital or digital/optical.

- Original Message -
From: "Wade Weppler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 1:24 PM
Subject: RE: [Asterisk-Users] Newbie Doubts


> There aren't any Ethernet-based channel banks available.  Channel Banks
> basically demux T1 digital info into 24 analog channels.  The Ethernet
jack
> on some channel banks is used for configuration and monitoring only.
>
> -wade
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Ricardo Saar Gemignani
> > Sent: Monday, July 07, 2003 12:44 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] Newbie Doubts
> >
> > Can´t I connect to the channel bank using ethernet? Why?
> >
> > - Original Message -
> > From: "Martin Pycko" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Monday, July 07, 2003 11:52 AM
> > Subject: Re: [Asterisk-Users] Newbie Doubts
> >
> >
> > > You plug a channel bank to a T1 in your PC connected either over T100P
> > or
> > > T400P.
> > >
> > > regards
> > > Martin
> > >
> > > On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote:
> > >
> > > > Hello everybody
> > > >
> > > > My doubt is about configuration. Can I use a channel bank like
> > zplex-10 or adtran, plug on it an T1, 24 POTS, an ethernet cable
connected
> > at the computer with Asterisk installed? Will it work? Will asterisk be
> > able
> > to control the system? Receive a call and work with all its
> > functions(transfer, conference, voice mail)?
> > > >
> > > > Thanks in advance,
> > > > Ricardo Saar Gemignani
> > > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
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>
>
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Re: [Asterisk-Users] Direct entry to your own voice mailbox

[Dan]

Thanks,

It seems to work with VoiceMailMain application too.

BR,
Dan

- Original Message - 
From: "Tan Aks" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 7:25 PM
Subject: Re: [Asterisk-Users] Direct entry to your own voice mailbox


> e.g.
>
> exten => 8501,1,VoiceMailMain2(${CALLERIDNUM})
>
> Tan
> telappliant.com
>
> - Original Message - 
> From: "Dan" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, July 07, 2003 4:47 PM
> Subject: [Asterisk-Users] Direct entry to your own voice mailbox
>
>
> Hi,
>
> There is any possibility to dial a specific extension and then enter in
your
> own mailbox (the one defined for that specific SIP phone) without asking
for
> the exxtension number but only for the password?
>
> I want to be the same extension for all phones, not a specific one for
each
> of them.
>
> It is possible to have a time stamp in the recorded message? I want to
know
> when the  message has been recorded.
>
> Thanks,
> Dan
>
>
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>
>
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[Asterisk-Users] System command..

[WipeOut .]

Can the system command be used to retrieve a variable from a mysql database using the 
mysql command line client??
or would it be simpler to write some sort of AGI type application??
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Re: [Asterisk-Users] Direct entry to your own voice mailbox

[Dan]

Hi,

Thank you very much for your help.


>
> > It is possible to have a time stamp in the recorded message? I want to
know
> > when the  message has been recorded.
>
> I think someone here was working on a patch for that, which was waiting
> for prompts to be recorded.  Not sure of the current status.
Why other prompts?
There is an application available (DateTime) to say current date and time.
It cannot be integrated in the Voicemail appication?

BR,
Dan



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[Asterisk-Users] Network design question

[Asterisk]

Hello!

My business is a wireless ISP.  I would like to offer voice to several 
business customers.  I have a * server, but still need some hardware cards 
for it.
I would like to provide individualized billing to these customers for their 
usage.  Most of these customers have Nortel PBX's.   Could I use a Cisco 
ATA  or a similar manufacturer to provide 1-2 line service?  I would like 
to get this set up correctly the first time.  What would you suggest or has 
someone already done this that could explain better?

Thanks,

Bill

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RE: [Asterisk-Users] Newbie Doubts

[Wade Weppler]

There aren't any Ethernet-based channel banks available.  Channel Banks
basically demux T1 digital info into 24 analog channels.  The Ethernet jack
on some channel banks is used for configuration and monitoring only.

-wade


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ricardo Saar Gemignani
> Sent: Monday, July 07, 2003 12:44 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Newbie Doubts
> 
> Can´t I connect to the channel bank using ethernet? Why?
> 
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, July 07, 2003 11:52 AM
> Subject: Re: [Asterisk-Users] Newbie Doubts
> 
> 
> > You plug a channel bank to a T1 in your PC connected either over T100P
> or
> > T400P.
> >
> > regards
> > Martin
> >
> > On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote:
> >
> > > Hello everybody
> > >
> > > My doubt is about configuration. Can I use a channel bank like
> zplex-10 or adtran, plug on it an T1, 24 POTS, an ethernet cable connected
> at the computer with Asterisk installed? Will it work? Will asterisk be
> able
> to control the system? Receive a call and work with all its
> functions(transfer, conference, voice mail)?
> > >
> > > Thanks in advance,
> > > Ricardo Saar Gemignani
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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RE: [Asterisk-Users] callgroup and pickupgroup

[Wade Weppler]

"group" has nothing to do with either callgroup or pickupgroup.

"callgroup" is a group of incoming "calls".
"pickupgroup" is for membership purposes.

A channel that belongs to a pickupgroup, can pickup all incoming calls on
the same callgroup by hitting *8#.

A channel can belong to multiple pickupgroup's, ie:

pickupgroup=1,2,3

This channel would then be able to pickup calls on callgroup 1, 2 or 3 by
hitting *8#.

-wade


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of carlos del mayor
> Sent: Monday, July 07, 2003 12:27 PM
> To: asterisk users
> Subject: [Asterisk-Users] callgroup and pickupgroup
> 
> Hi,
> I asked a time ago what were callgroup and pickup
> group used for. I have done some proofs and all, and
> I'm not sure if I have pick the idea up well!!
> 
> That's what I understand:
> For example: group=1 callgroup =2 and pickupgroup=2
> and my phone is a membership of the group 1.
> that's mean that when a phone that belong to group 2
> is ringing, I'll be able to answer this call dialing
> *8#. I hope I'm not wrong in this.
> 
> My question is, why I need the two keywords, callgroup
> and pickupgroup, if they're always the same and mean
> the same, don't they?
> 
> Thanks a lot for clarifying me this!!!
> cmayor
> 
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RE: [Asterisk-Users] Newbie Doubts

[Joe Antkowiak]

Because a channel bank converts a DS1 into 24 DS0s and vice versa.  Your
channel bank may have an Ethernet port on it, but it is just for management.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Saar
Gemignani
Sent: Monday, July 07, 2003 12:44 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie Doubts

Can´t I connect to the channel bank using ethernet? Why?

- Original Message -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 11:52 AM
Subject: Re: [Asterisk-Users] Newbie Doubts


> You plug a channel bank to a T1 in your PC connected either over T100P or
> T400P.
>
> regards
> Martin
>
> On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote:
>
> > Hello everybody
> >
> > My doubt is about configuration. Can I use a channel bank like
zplex-10 or adtran, plug on it an T1, 24 POTS, an ethernet cable connected
at the computer with Asterisk installed? Will it work? Will asterisk be able
to control the system? Receive a call and work with all its
functions(transfer, conference, voice mail)?
> >
> > Thanks in advance,
> > Ricardo Saar Gemignani
> >
>
> ___
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>

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Re: [Asterisk-Users] Asterisk and VMWare

[WipeOut .]

If you want to use it for IP only it __MAY__ work but probably not all that well 
unless you have a really strong processor..

VMWare abstracts the hardware layer you will not be able to get it to communicate 
fully with any cards in order to connect phone lines..

> Hi,
> 
> There is any experience using Asterisk with VMWare?
> I think about installing a virtual linux box over VMWare and then Asterisk
> over it.
> 
> Thanks,
> Dan
> 
> 
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Re: [Asterisk-Users] Newbie Doubts

[Ricardo Saar Gemignani]

Can´t I connect to the channel bank using ethernet? Why?

- Original Message -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 11:52 AM
Subject: Re: [Asterisk-Users] Newbie Doubts


> You plug a channel bank to a T1 in your PC connected either over T100P or
> T400P.
>
> regards
> Martin
>
> On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote:
>
> > Hello everybody
> >
> > My doubt is about configuration. Can I use a channel bank like
zplex-10 or adtran, plug on it an T1, 24 POTS, an ethernet cable connected
at the computer with Asterisk installed? Will it work? Will asterisk be able
to control the system? Receive a call and work with all its
functions(transfer, conference, voice mail)?
> >
> > Thanks in advance,
> > Ricardo Saar Gemignani
> >
>
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[Asterisk-Users] register line on sip.conf

[carlos del mayor]

Hi,

Only one question: Is this the generic format for a
register line on sip.conf?
[EMAIL PROTECTED]:[EMAIL PROTECTED]:port/exten
 
Or better this one?

user:[EMAIL PROTECTED]

Or if there is another one, please tell me!
Thans a lot in advance!
cmayor

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Re: [Asterisk-Users] Direct entry to your own voice mailbox

[The Traveller]

Hey Dan,

On Mon, Jul 07, 2003 at 18:47:07 +0300, Dan wrote:

> Hi,
> 
> There is any possibility to dial a specific extension and then enter in your
> own mailbox (the one defined for that specific SIP phone) without asking for
> the exxtension number but only for the password?

Sure.  Pass the mailbox as an arg to the "VoiceMailMain" or
"VoiceMailMain2"-app.  See "show application voicemailmain" on the
Asterisk-console for more info.

> I want to be the same extension for all phones, not a specific one for each
> of them.

I think you could do that by passing "${CALLERIDNUM}" as the arg.

> It is possible to have a time stamp in the recorded message? I want to know
> when the  message has been recorded.

I think someone here was working on a patch for that, which was waiting
for prompts to be recorded.  Not sure of the current status.



   Grtz,

  Oliver
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Re: [Asterisk-Users] FWD trouble - 407 error

[Iain Stevenson]

Thanks for that.  It seems one now needs something like this in sip.conf:

[fwd.pulver.com]
type=peer
host=fwd.pulver.com
username=12345
secret=mysecret
fromdomain=fwd.pulver.com
callerid="Free World Dialup"
All is well again ...

 Iain



--On Saturday, July 5, 2003 9:31 pm -0400 "James H. Cloos Jr." 
<[EMAIL PROTECTED]> wrote:

"Iain" == Iain Stevenson <[EMAIL PROTECTED]> writes:
Iain> I didn't used to have any trouble with FWD and * is registering
Iain> with FWD OK.  Has FWD changed or * changed in a way that might
Iain> cause this error?
Jeff just announce an upgrade to fwd the other day.

One change is that callers have to be logged in.  (This is to help
ensure the cnid info send in the sip invite accurately reflects the
callers fwd number.)  This change was added after a rash of nuisance
calls?.
-JimC

? As a disclaimer:  I started a thread on the fwd list about
  these after I received a couple of mostly amusing ones; the
  discussion on that thread may have led to the change.
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[Asterisk-Users] callgroup and pickupgroup

[carlos del mayor]

Hi,
I asked a time ago what were callgroup and pickup
group used for. I have done some proofs and all, and
I'm not sure if I have pick the idea up well!!

That's what I understand: 
For example: group=1 callgroup =2 and pickupgroup=2
and my phone is a membership of the group 1.
that's mean that when a phone that belong to group 2
is ringing, I'll be able to answer this call dialing
*8#. I hope I'm not wrong in this. 

My question is, why I need the two keywords, callgroup
and pickupgroup, if they're always the same and mean
the same, don't they?

Thanks a lot for clarifying me this!!!
cmayor

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Re: [Asterisk-Users] Direct entry to your own voice mailbox

[Tan Aks]

e.g.

exten => 8501,1,VoiceMailMain2(${CALLERIDNUM})

Tan
telappliant.com

- Original Message - 
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 4:47 PM
Subject: [Asterisk-Users] Direct entry to your own voice mailbox


Hi,

There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?

I want to be the same extension for all phones, not a specific one for each
of them.

It is possible to have a time stamp in the recorded message? I want to know
when the  message has been recorded.

Thanks,
Dan


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[Asterisk-Users] modules.conf

[carlos del mayor]

Hi everybody,

I have two little question about modules.conf. 

1)As I have seen, to make Asterisk load chan_capi.so
and chan_modem.so you must have: load=>chan_capi.so
and load => chan_modem.so in your modules.conf. But I
had understood some time ago that setting autoload =>
yes made Asterisk load every module that was necesary.
Then, why must I load these channels explicitely?

2)For what is used the section [global]?

thanks a lot
cmayor

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[Asterisk-Users] Asterisk and VMWare

[Dan]

Hi,

There is any experience using Asterisk with VMWare?
I think about installing a virtual linux box over VMWare and then Asterisk
over it.

Thanks,
Dan


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[Asterisk-Users] Initiations in IP voice/Hybrid Voice/etc...

[Xisco]

Hi everybody,
 
From now I have been working with IVR feature of *. 

 
But now We want to began to use VoIP, SIP, IAX, 
etc. Now we have severals initials scenarios where we need to know which 
tecnology must apply in order to have a good application. So If somebody can 
give us the tecnology and some start-idea we will be very 
pleasured.
 
Scenario 1. (The two phones are IP 
phones).
 
IP Phone A Asterisk--IP Cloud 
public/privateAsteriskIP Phone B.
 
Scenario 2. (One IP phone and one ISDN 
phone)
 
ISDN Phone ---ISDN Public 
CloudAsteriskIP Cloud public/private--Asterisk---IP 
Phone.
 
Scenario 3. (Two ISDN phones).
 
ISDN Phone ---ISDN Public 
CloudAsteriskIP Cloud 
public/private--Asterisk---ISDN Public CloudISDN 
Phone
 
I thing that all the scenarios are avaible with 
asterisk, but I don't know which protocol will give me better results (IAX, SIP, 
H.323). May be I will need to apply QoS in IP cloud, so it doesn't have 
relevance in VoIP protocol.
 
Thks a lot tto everybody in the list.
 

Re: [Asterisk-Users] Problems with TDM40P

[The Traveller]

Hey Adam,

On Tue, Jul 08, 2003 at 00:58:08 +1000, Adam Goryachev wrote:

> Without quite just saying me too, see below...

[...]

> > My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected
> > to an analog line to my telco, and a TDM40P with analog phones
> 
> I have an AMD XP 1800 with 256MB RAM, and a single IDE HDD. I have a X100P,
> a TDM40P and a ISDN BRI card using ISDN4Linux driver.
> 
> [EMAIL PROTECTED] root]# cat /proc/interrupts
>CPU0
>   0:1929531  XT-PIC  timer
>   1:  2  XT-PIC  keyboard
>   2:  0  XT-PIC  cascade
>   5:  0  XT-PIC  usb-uhci, usb-uhci
>   8:  1  XT-PIC  rtc
>  10:   18895682  XT-PIC  eth0, wcfxo
>  11:  23378  XT-PIC  HiSax
>  12:   17057246  XT-PIC  wcfxs
>  14:  45267  XT-PIC  ide0
>  15: 13  XT-PIC  ide1
> NMI:  0
> ERR:  2

I actually also have an ISDN BRI-card in there, but it was not in use
during these tests.

> > My problems with this setup are as follows:
> >
> > - I hear a soft fading hiss in the background on the TDM40P-connected
> > phones.  It's also present when I call something like voicemail
> > or a conference on my local box, so it doesn't seem to come from
> > the X100P.  If I uncomment the "NO_CALIBRATION"-option in the Zaptel
> > Makefile, the hiss decreases in volume, but is still audible.
> 
> I have the same problem, but it seems to only be on one phone. Originally I
> had all 4 extensions passed across a single CAT5 cable, now I have 4
> separate CAT5 cables with RJ45 connectors on them and it seems better, but
> it is still there. I also get a funny 'ring' tone if I am in

Currently, some of the lines run over the same CAT5-wire for me as well,
but I also have some phones which are connected directly to the TDM40P,
with at most 2 meters of wire.  They have the same problems, though.

> voicemail/conference/phone call/whatever and another extension rings (sort
> of like a crossed line perhaps, but it isn't the sound of the ring, perhaps
> the actual ring voltage being applied/leaked to my line).

I've noticed this leak as well.  Besides that, I can also clearly hear
the caller-ID and MWI-status beeing signalled to other extensions in
the background.  It doesn't seem to be my wiring, as I also hear it
with 2 phones connected directly to the TDM40P, over seperate cables.

> > - Some of my TDM40P-connected phones go dead frequently.  By "dead",
> > I mean that the line-voltage goes away, so there's nothing there when
> > they're picked up.  When this happens, the remedy is to call that
> > extension and let it ring once, or restart Asterisk, after which the
> > line-voltage re-appears.  I have this problem with several different
> > phones and know at least 1 other person with similar problems.
> 
> I have the same problem, usually all 4 lines will go at the same time, but
> while I was re-cabling (lots of pickup, listen for dialtone, hangup cycles
> on each line to see if my cabling was working) and I got this a number of
> times. ie, one or more extensions not working. My fix was a stop asterisk,
> remove modules, insert modules, start asterisk.

Yups, I certainly noticed that lots of unplugging / reconnecting of
phones also seems to trigger it.

> I've also had a number of problems where all TDM400 extensions are dead but
> the X100P and ISDN still work fine. When I stop asterisk, and remove the
> modules, all is OK, but as soon as I insert the modules, the machine locks
> up, and a power off/on is the only thing that works. I don't really know how
> to track this one down, or how to diagnose it to better assist other people
> to fix the problem

That's intresting.  I'm seeing that one as well.  About 1 in 10 to 20 times
I unload and load the Zaptel-drivers, "wcfxs" prints some sort of register-
dump and informs me that the Freshmaker register-test failed, after which
the machine is locked up solid.  The register-dump contains mostly "ff"'s
in such a case and there doesn't seem to be a kernel-panic.

I suspect the "wcfxs" driver-init of the TDM40P does some pretty nasty
stuff, as even during a regular and succesfull init, the machine becomes
completely unresponsive for the duration of the init.  I suspect this
frozen state doesn't get undone properly in some failure-situations,
which leaves you with a frozen machine if that happens.

> > - Making data-calls from a line on the TDM40P, through the X100P, to
> > an external number is almost impossible.  Even 2400bps is unreliable,
> 
> I've not tried data calls, but I think there is a d parameter to the Dial
> application (see show application dial) which assists with data calls. Are
> you using that?

Yups, I've added the "d", but it doesn't appear to make any difference.

Thanks for your information.  Hopefully, Mark will be able to shed
some light on it as well.  I'm also available for supplying more
debug-info, if needed.

[Asterisk-Users] Direct entry to your own voice mailbox

[Dan]

Hi,

There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?

I want to be the same extension for all phones, not a specific one for each
of them.

It is possible to have a time stamp in the recorded message? I want to know
when the  message has been recorded.

Thanks,
Dan


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RE: [Asterisk-Users] How to make * send RTCP reports

[HT]

Thank you for the answer.

Anyone working on that?

I am trying in the meantime to disable the RTCP reports on the gateways,
hoping that it will work like that.

hristo

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: Saturday, July 05, 2003 9:52 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to make * send RTCP reports

> I can see on the console that * is detecting incoming RTCP reports so
there
> should be some RTCP functionality in it (although I have seen a message
from
> February saying the opposite). My question is if/how can I make * send
RTCP
> report to the vocaltec gateways. I think any RTCP packet will do the trick
> as long as the vocaltec gateway gets it on a regular basis (I don't care
if
> the information in it is correct).

It would require some additional code, but could be done.  Some of the
basics are in place now but not nearly enough code, as it did not end up
being required for video as we had though tit would be.

Mark

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[Asterisk-Users] three way calling and cisco ata 186

[Pavel Zheltouhov]

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk  as pbx. I need feature called as 'three way calling' or 
'transfer with consultation'. Registering,calling and 'blind transfer' 
work fine.

Is this feature provided by sip clients or by asterisk itself ?
What I have to configure in ATA   and what keys I have to press
on my phones ?
Three way calling is enabled by setting  bits 28-29 to 10
 ( Select the Cisco VG248 Style for mid-call services.
 as described at 
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a00800c4d1f.html#42433
But it seems not works,I always get conference call with 3 persons.

--
Pavel Zheltouhov
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Re: [Asterisk-Users] Newbie Doubts

[Martin Pycko]

You plug a channel bank to a T1 in your PC connected either over T100P or
T400P.

regards
Martin

On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote:

> Hello everybody
>
> My doubt is about configuration. Can I use a channel bank like zplex-10 or 
> adtran, plug on it an T1, 24 POTS, an ethernet cable connected at the computer with 
> Asterisk installed? Will it work? Will asterisk be able to control the system? 
> Receive a call and work with all its functions(transfer, conference, voice mail)?
>
> Thanks in advance,
> Ricardo Saar Gemignani
>

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RE: [Asterisk-Users] Problems with TDM40P

[Adam Goryachev]

Without quite just saying me too, see below...

> I'm experiencing some problems with a TDM40P and was wondering if
> anyone else on this list has similar experiences, or maybe even
> a possible solution.

Similar experiences...

> My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected
> to an analog line to my telco, and a TDM40P with analog phones

I have an AMD XP 1800 with 256MB RAM, and a single IDE HDD. I have a X100P,
a TDM40P and a ISDN BRI card using ISDN4Linux driver.

[EMAIL PROTECTED] root]# cat /proc/interrupts
   CPU0
  0:1929531  XT-PIC  timer
  1:  2  XT-PIC  keyboard
  2:  0  XT-PIC  cascade
  5:  0  XT-PIC  usb-uhci, usb-uhci
  8:  1  XT-PIC  rtc
 10:   18895682  XT-PIC  eth0, wcfxo
 11:  23378  XT-PIC  HiSax
 12:   17057246  XT-PIC  wcfxs
 14:  45267  XT-PIC  ide0
 15: 13  XT-PIC  ide1
NMI:  0
ERR:  2

> My problems with this setup are as follows:
>
> - I hear a soft fading hiss in the background on the TDM40P-connected
> phones.  It's also present when I call something like voicemail
> or a conference on my local box, so it doesn't seem to come from
> the X100P.  If I uncomment the "NO_CALIBRATION"-option in the Zaptel
> Makefile, the hiss decreases in volume, but is still audible.

I have the same problem, but it seems to only be on one phone. Originally I
had all 4 extensions passed across a single CAT5 cable, now I have 4
separate CAT5 cables with RJ45 connectors on them and it seems better, but
it is still there. I also get a funny 'ring' tone if I am in
voicemail/conference/phone call/whatever and another extension rings (sort
of like a crossed line perhaps, but it isn't the sound of the ring, perhaps
the actual ring voltage being applied/leaked to my line).

> - Some of my TDM40P-connected phones go dead frequently.  By "dead",
> I mean that the line-voltage goes away, so there's nothing there when
> they're picked up.  When this happens, the remedy is to call that
> extension and let it ring once, or restart Asterisk, after which the
> line-voltage re-appears.  I have this problem with several different
> phones and know at least 1 other person with similar problems.

I have the same problem, usually all 4 lines will go at the same time, but
while I was re-cabling (lots of pickup, listen for dialtone, hangup cycles
on each line to see if my cabling was working) and I got this a number of
times. ie, one or more extensions not working. My fix was a stop asterisk,
remove modules, insert modules, start asterisk.

I've also had a number of problems where all TDM400 extensions are dead but
the X100P and ISDN still work fine. When I stop asterisk, and remove the
modules, all is OK, but as soon as I insert the modules, the machine locks
up, and a power off/on is the only thing that works. I don't really know how
to track this one down, or how to diagnose it to better assist other people
to fix the problem

> - Making data-calls from a line on the TDM40P, through the X100P, to
> an external number is almost impossible.  Even 2400bps is unreliable,

I've not tried data calls, but I think there is a d parameter to the Dial
application (see show application dial) which assists with data calls. Are
you using that?

Regards,
Adam

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[Asterisk-Users] Newbie Doubts

[Ricardo Saar Gemignani]

Hello everybody
 
    My doubt is about configuration. 
Can I use a channel bank like zplex-10 or adtran, plug on it an T1, 24 
POTS, an ethernet cable connected at the computer with Asterisk installed? 
Will it work? Will asterisk be able to control the system? Receive a call and 
work with all its functions(transfer, conference, voice 
mail)? 
 
Thanks in advance,
Ricardo Saar 
Gemignani 

Re: [Asterisk-Users] ZapRas: Anyone has an idea how to configure it???

[Mark Spencer]

>   Is there anybody out there that has an idea how to make Asterisk
> work with ZapRAS with an E1 interface? In addition, I would like to know
> if anybody ever tried making GSM phones to connect to asterisk via PPP,
> over an ISDN v.110 connection or v.120 connection.

There are at least two people running this.  Documentation is fairly
sparse, but basically you run ZapRas with patched pppd (patches available
at ftp.digium.com).  Please remember this is ISDN RAS *only* and has no
modulation in it.

Mark

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[Asterisk-Users] Problems with TDM40P

[The Traveller]

Heya all,

I'm experiencing some problems with a TDM40P and was wondering if
anyone else on this list has similar experiences, or maybe even
a possible solution.

My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected
to an analog line to my telco, and a TDM40P with analog phones
connected.  The TDM-card is not sharing any interrupts, but the
X100P is, with the 2 Adaptec SCSI hostadapters on the mainboard.
There's only a tapestreamer on one of them, though, and it's mostly
idle, as I only use it during backups.  Not loading the SCSI-drivers
at all doesn't make any difference in the problems below.
All of the disks in the machine are IDE, and those controllers have their
own interrupts.

My problems with this setup are as follows:

- I hear a soft fading hiss in the background on the TDM40P-connected
phones.  It's also present when I call something like voicemail
or a conference on my local box, so it doesn't seem to come from
the X100P.  If I uncomment the "NO_CALIBRATION"-option in the Zaptel
Makefile, the hiss decreases in volume, but is still audible.

- Some of my TDM40P-connected phones go dead frequently.  By "dead",
I mean that the line-voltage goes away, so there's nothing there when
they're picked up.  When this happens, the remedy is to call that
extension and let it ring once, or restart Asterisk, after which the
line-voltage re-appears.  I have this problem with several different
phones and know at least 1 other person with similar problems.

- Making data-calls from a line on the TDM40P, through the X100P, to
an external number is almost impossible.  Even 2400bps is unreliable,
and anything higher than that usually doesn't even connect.  Voice-
calls sound very clear, though.  I suspect it might have something
to do with the first problem above, as I've also tried data-calls
between ports on that same TDM40P, with the exact same results.  I've
even tried a completely different TDM40P, so it's unlikely to be
a hardware-problem with that.  The afforementioned "NO_CALIBRATION"-
option seems to make a slight difference in the quality of data-calls,
but not much.



   Grtz,

  Oliver
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Re: [Asterisk-Users] Voicemail2 Contexts

[Mark Spencer]

VoicemailMain2(@context)

Mark

On Mon, 7 Jul 2003, WipeOut . wrote:

> Hi,
>
> Can someone explain how or if voicemail2 contexts relate to the dialplan??
>
> If I had the following in voicemail.conf..
>
> [default]
> 1234 => 4567,Company1..
>
> [company2]
> 1234 => ,Company2..
>
> [company3]
> 1234 => 9876,Company3..
>
> How would I make sure that a user from Company2 did not get prompted for the 
> password for mailbox 1234 in the company3 or default  voicemail contexts??
>
> Thanks
>
>
> --
> __
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RE: [Asterisk-Users] Accurate Billing

[Matteo Brancaleoni]

Hi.

First of all : please disable html. Not everyone is able
to read html email. Also mailing lists netiquette expects
that a members writes only in plain text.

Second one:
cdr -provides- accurate info on a call. every call, estabilished
or not, is logged, the is up to you to get what u need.
The 'billable' time is logged as cdr->billsec and is the
real duration of the call between the answer and hungup.
If you call a sip phone, for example, let it rings for
5 secs , then answer ad speak for 10 secs, you'll have:
cdr->duration : 15 secs
cdr->billsec  : 10 secs
so 5 secs is ringing time, call setup and so on...

That's valid on all digital interfaces (even i4l or capi),
since the answer/hangup/ringing and so on are signalled
with proper messages digitally.

Unfortunately, when using analog interfaces, that's not
true, since there's no easy way to decide that a call
has been answered, since our channel is 'answered' as soon as
we connect to the analog line. This is for fxo (phone lines),
for fxs (phones) we should have a 100% accurate signalling,
since on fxs we know when the user switched to off hook.

Mark has written a call progress detector for analog lines
(callprogress=yes in zapata.conf), but is entirely dsp-based,
experimental and US only ;)

Also dsp detection isn't 100% reliable.

So, if you're using digital lines (e1,t1,bri) you're set.
the cdr provides everything you need for billing.

If you're using analog lines, you're in trouble ;)
but a real provider that wants to 'provide' phone
services , should go in the digital world ;)

just my 2 cents

Matteo.

Scrive [EMAIL PROTECTED]:

> hi,
> r u saying that, if i use E100P or E400P interface, and if i make a call, can
> i differenciate between a answered and non answered call, and can i bill only
> to the answered call?
> Surajee
> On Sun, 2003-07-06 at 23:17, Kim C. Callis wrote:
> > Steve,
> > 
> > What exactly would be classified as a digital ZAP device?
> 
> T1/E1 interfaces, so T100P, E100P, T400P, E400P
> 
> If you need to see examples, I could probably dig up CDR records where
> busy is indicated, and where no answer is indicated and there is a
> definate difference between call duration and stop-start duration. 
> 
> > > -Original Message-
> > > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Steven Critchfield
> > > Sent: Sunday, July 06, 2003 8:58 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: Re: [Asterisk-Users] Accurate Billing
> > > 
> > > On Sun, 2003-07-06 at 22:07, [EMAIL PROTECTED] wrote:
> > > > hi everyone,
> > > >
> > > > I know this issue has been raised many times before, i think still
> > the
> > > > problem remains. When a call is made through a Zap channel, whether
> > it
> > > > is actually made or not (irrespective of whether, engaged, busy, or
> > > > actually answered), asterisk logs it in CDRs as a call made. This
> > > > makes it impossible to do an accurate billing. Has anybody found a
> > way
> > > > to overcome this problem, if yes, please let me/us know.
> > > 
> > > If you are on a digital Zap interface, then it is known. If you are on
> > > an analog interface, then there is no way to know the other answered
> > or
> > > not.
> > > 
> > > --
> > > Steven Critchfield <[EMAIL PROTECTED]>
> &!
> gt; > 
> > > ___
> 
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> Steven Critchfield <[EMAIL PROTECTED]>
> 
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> 
> 
> 
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-- 

Matteo Brancaleoni
Espia System Administrator
http://www.espia.it

-
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RE: [Asterisk-Users] Accurate Billing

[Steven Critchfield]

On Mon, 2003-07-07 at 01:15, [EMAIL PROTECTED] wrote:
> hi,
> 
> r u saying that, if i use E100P or E400P interface, and if i make a
> call, can i differenciate between a answered and non answered call,
> and can i bill only to the answered call?

You would have to use a PRI(PRA?) line attached to the E(1|4)00P.


> On Sun, 2003-07-06 at 23:17, Kim C. Callis wrote:
> > Steve,
> > 
> > What exactly would be classified as a digital ZAP device?
> 
> T1/E1 interfaces, so T100P, E100P, T400P, E400P
> 
> If you need to see examples, I could probably dig up CDR records where
> busy is indicated, and where no answer is indicated and there is a
> definate difference between call duration and stop-start duration. 
> 
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Steven Critchfield
> > > Sent: Sunday, July 06, 2003 8:58 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: Re: [Asterisk-Users] Accurate Billing
> > > 
> > > On Sun, 2003-07-06 at 22:07, [EMAIL PROTECTED] wrote:
> > > > hi everyone,
> > > >
> > > > I know this issue has been raised many times before, i think
> still
> > the
> > > > problem remains. When a call is made through a Zap channel,
> whether
> > it
> > > > is actually made or not (irrespective of whether, engaged, busy,
> or
> > > > actually answered), asterisk logs it in CDRs as a call made.
> This
> > > > makes it impossible to do an accurate billing. Has anybody found
> a
> > way
> > > > to overcome this problem, if yes, please let me/us know.
> > > 
> > > If you are on a digital Zap interface, then it is known. If you
> are on
> > > an analog interface, then there is no way to know the other
> answered
> > or
> > > not.
> > > 
> > > --
> > > Steven Critchfield <[EMAIL PROTECTED]>
> &! gt; > 
> > > ___ 
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> Steven Critchfield <[EMAIL PROTECTED]>
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/li! stinfo/asterisk-users
> 
> 
> 
> 
> --This mail sent through OmniBIS.com--
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-- 
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Re: [Asterisk-Users] ZapRas: Anyone has an idea how to configure it???

[Thilo Salmon]

On Mon, 2003-07-07 at 12:49, Lord Stroud wrote:
>   Is there anybody out there that has an idea how to make Asterisk
> work with ZapRAS with an E1 interface? In addition, I would like to know
> if anybody ever tried making GSM phones to connect to asterisk via PPP,
> over an ISDN v.110 connection or v.120 connection.

Won't work - there is no support for v.110 and v.120.

For ISDN clear channel access make sure pppd resides in /usr/sbin,
configure /etc/ppp/options and make asterisk start ZapRAS on your
favourite extension.

Thilo


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Re: [Asterisk-Users] Virtual fax on the Asterisk box

[marrandy]

On Monday 07 July 2003 08:24 am, Dan wrote:
> Hi Andrea,
> 
> This is very interesting starting point.
> 
> Thanks,
> Dan


Great going Dan,

You posted 6 new lines, left an additional 134 lines of old junk, 4 original 
headers and 5 old footers.

Can people trim their posts when they reply.

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[Asterisk-Users] problems with new FXS module

[Paulo Mannheimer]

HI, I just installed an additional module to a TDM400P card.
Everything worked fine for a while, but after rebooting the server the new
channel fails to be recognized by asterisk. Here is the output

 

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

    --
Registered channel 1, FXS Kewlstart signalling

    --
Registered channel 2, FXO Kewlstart signalling

    --
Registered channel 3, FXO Kewlstart signalling

WARNING[1024]: File chan_zap.c, Line 576 (zt_open):
Unable to specify channel 4: No such device

ERROR[1024]: File chan_zap.c, Line 4746 (mkintf):
Unable to open channel 4: No such device

here = 0, tmp->channel = 0, channel = 4

ERROR[1024]: File chan_zap.c, Line 6404 (load_module):
Unable to register channel '2-4'

WARNING[1024]: File loader.c, Line 299 (ast_load_resource):
chan_zap.so: load_module
failed, returning -1

WARNING[1024]: File loader.c, Line 394 (load_modules):
Loading module chan_zap.so failed!

 

 

my /etc/zaptel.conf

fxsks=1

fxoks=2-4

loadzone = us

defaultzone=us

 

 

my /etc/asterisk/Zapata.conf

signalling=fxs_ks

channel=1

signalling=fxo_ks

channel=2-4

 

Here is the output for cat /proc/zaptel/*

Span 1: WCFXO/0 "Wildcard X101P Board 1"

    1
WCFXO/0/0 FXSKS

Span 2: WCFXS/0 "Wildcard S400P Prototype Board 1"

    2
WCFXS/0/0 FXOKS

    3
WCFXS/0/1 FXOKS

    4
WCFXS/0/2 FXOKS

    5 WCFXS/0/3

 

Any hints?

 

Re: [Asterisk-Users] Virtual fax on the Asterisk box

[Dan]

Hi Andrea,

This is very interesting starting point.

Thanks,
Dan

- Original Message - 
From: "Andrea Venturi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 2:44 PM
Subject: Re: [Asterisk-Users] Virtual fax on the Asterisk box


> > - Original Message - 
> > From: "Dan" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Monday, July 07, 2003 10:13 AM
> > Subject: Re: [Asterisk-Users] Virtual fax on the Asterisk box
> >
> >
> > Hi,
> >
> > What I need is a pure software solution, to avoid any other hardware to
get
> > that functionality.
> >
>
> i recall t38modem (a soft modem working as an h323 endpoint)
>
>http://www.openh323.org/t38.html
>
> it should/could work against an asterisk gateway with h323 support (any
> flavour..) and it should expose a serial device driven with AT commands
> and acting as a soft-modem for just sending (and receive?!) fax ( i
> don't remember if it's just class 1 or 2 or whatever..)
>
> BTW i didn't have the time to try it..
>
> anyone with some experience to share?
>
> bye
>
> > Thanks,
> > Dan
> >
> >
> > - Original Message - 
> > From: "Simon Woodhead" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Monday, July 07, 2003 11:50 AM
> > Subject: Re: [Asterisk-Users] Virtual fax on the Asterisk box
> >
> >
> >
> >>Hylafax Dan. It isn't that elegant though as you'll need to wire an
> >
> > analogue
> >
> >>port to each fax/modem. AFAIK there isn't a virtual fax/modem provided
by
> >
> > *
> >
> >>that another programme can use.
> >>
> >>W
> >>
> >>- Original Message - 
> >>From: "Dan" <[EMAIL PROTECTED]>
> >>To: <[EMAIL PROTECTED]>
> >>Sent: Saturday, July 05, 2003 7:20 AM
> >>Subject: [Asterisk-Users] Virtual fax on the Asterisk box
> >>
> >>
> >>Hi all,
> >>
> >>I want to get the following functionality: define one extension as a
> >
> > virtual
> >
> >>fax machine.
> >>Every fax redirected to that extension to be converted in a picture file
> >>(bmp/jpg/gif or something else) and then attached to an email and send
to
> >
> > an
> >
> >>e-mail address.
> >>Are you aware of a linux based application who does something like this
> >
> > and
> >
> >>can be installed on the same computer as Asterisk?
> >>Another possibility is a Windows based software SIP fax which can be
> >>registered with Asterisk.
> >>
> >>Thanks,
> >>Dan
> >>
> >>
> >>___
> >>Asterisk-Users mailing list
> >>[EMAIL PROTECTED]
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>___
> >>Asterisk-Users mailing list
> >>[EMAIL PROTECTED]
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> ___
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> [EMAIL PROTECTED]
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>


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Re: [Asterisk-Users] Virtual fax on the Asterisk box

[Dave Weis]

On Mon, 7 Jul 2003, Andrea Venturi wrote:
> i recall t38modem (a soft modem working as an h323 endpoint)
>http://www.openh323.org/t38.html
> anyone with some experience to share?

I tried it a few months ago with a maxtnt and a couple t1's. It would
usually receive one page but die on the beginning of the second. I worked
with the developer for a while but he wasn't sure what was causing
it. It's probably improved by now, however.

-- 
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
  encroachments of those in power than by violent 
  and sudden usurpations."- James Madison

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Re: [Asterisk-Users] Remote * Using IAX

[WipeOut .]

Setup all your phones to connect to the closest * server..

Setup IAX between the two servers..

Use the switch command on each server and point it to the other server so that 
extentions will be available to both..

You will only need to "register" one server with the other if it have a dynamic IP 
address..

Hope that helps..

> 
> I need to configure an * box to connect to a primary * box which is attached
> to a PRI in order to make calls using both the same E1 connection.
> 
> I know the best solution is to use IAX (altough i could connect the IP
> phones on the remote site directly to the main * box that is VPNed with the
> remote one).
> 
> I've some doubt on how to config IAX to work in this situation.
> 
> I think that:
> 
> On the main * box I'll have to configure all the users of the remote site as
> IAX users (phone1, phone2, ecc.).
> 
> On the remote * i need to set up a "register =>" statement in order to have
> the remote * connected to the first one.
> 
> Is this right?
> 
> In other words, the remote should register on the main * box, but I'm not
> sure of where i've to put the phones (all SIP GrandStream phones) to be
> registered. In the remote or in the main * IAX.conf file? (or both, maybe).
> 
> Any help would be appreciated, since I've not been able to understand on my
> own even reading the whole lot of messages about IAX on this list :(
> 
> -- 
> Stefano Finetti
> Technical Coordinator
> Lynx Automotive srl
> [EMAIL PROTECTED]
> Tel: 199 79 79 30
> Fax: 06 233 227 934
> Linux Registered User #271978
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [Asterisk-Users] Accurate Billing

[Xisco]

Hi everybody,

I have the same problem severals month ago, and the only solution that I
found was modify code on app_dial.c in order to return severals values:

-Return the time when the call have been established (time() unix time),
so you have to see this value and calculate time just before dial. In order
to get the seconds if the call is established.
-And return 0 or -1 if the call hasn't established.

I use AGI scripting with this modification.

Bye.


- Original Message - 
From: "marrandy" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 1:46 PM
Subject: Re: [Asterisk-Users] Accurate Billing


> On Sunday 06 July 2003 11:07 pm, [EMAIL PROTECTED] wrote:
> > hi everyone,
> > I know this issue has been raised many times before, i think still
the
> problem remains. When a call is made through a Zap channel, whether it is
> actually made or not (irrespective of whether, engaged, busy, or actually
> answered), asterisk logs it in CDRs as a call made. This makes it
impossible
> to do an accurate billing. Has anybody found a way to overcome this
problem,
> if yes, please let me/us know.
> > Thank you inadvance,
> > Surajee
> >
>
> Do you know your sending in HTML ???
>
> -- 
> Claret is the liquor for boys; port for men; but he who aspires to be a
hero
> ... must drink brandy.
> -- Samuel Johnson
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [Asterisk-Users] IAX Bandwidth Question

[WipeOut .]

I have 2 asterisk systems connected by a 56kbps internet dialup (so at best 33.6k in 
both directions) using IAX and GSM...

The one * box is at my home and the other is in the office (before anyone freaks this 
is a test environment)... Provided nothing is using the line at the same time I am 
able to carry 1 voice call over the dialup link with no real latency.. Ping times when 
no call is in session are about 165-175ms and when a call is active 190-240ms.. I just 
ran a call for 5 mins to see if the ping time climbed as you are experiencing and it 
didn't increase at all..

So you may have some other issue causing the increase in ping times..

Sorry I probably wasn't much help..

> Hi,
> 
> I am using IAX to communicate between 2 sites, each site is using a 256k/64k ADSL 
> Connection.
> 
> 
> I have noticed that when I connect my ping time to my 1st hop jumps from 30~ms to 
> over 12,000ms in over a period of about 10 minutes, it just keeps climbing until the 
> link is saturated.
> 
> Naturally, there is a very long delay when speaking.
> 
> What bandwidth would be adequate for IAX? or how can I tune my config to work better 
> with my current bandwidth situation.
> I am using GSM codec and bandwidth=low in iax.conf
> 
> Thanks in advance.
> 
Jay.
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Re: [Asterisk-Users] Virtual fax on the Asterisk box

[Andrea Venturi]

- Original Message - 
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 10:13 AM
Subject: Re: [Asterisk-Users] Virtual fax on the Asterisk box

Hi,

What I need is a pure software solution, to avoid any other hardware to get
that functionality.
i recall t38modem (a soft modem working as an h323 endpoint)

  http://www.openh323.org/t38.html

it should/could work against an asterisk gateway with h323 support (any 
flavour..) and it should expose a serial device driven with AT commands 
and acting as a soft-modem for just sending (and receive?!) fax ( i 
don't remember if it's just class 1 or 2 or whatever..)

BTW i didn't have the time to try it..

anyone with some experience to share?

bye

Thanks,
Dan
- Original Message - 
From: "Simon Woodhead" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 11:50 AM
Subject: Re: [Asterisk-Users] Virtual fax on the Asterisk box



Hylafax Dan. It isn't that elegant though as you'll need to wire an
analogue

port to each fax/modem. AFAIK there isn't a virtual fax/modem provided by
*

that another programme can use.

W

- Original Message - 
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, July 05, 2003 7:20 AM
Subject: [Asterisk-Users] Virtual fax on the Asterisk box

Hi all,

I want to get the following functionality: define one extension as a
virtual

fax machine.
Every fax redirected to that extension to be converted in a picture file
(bmp/jpg/gif or something else) and then attached to an email and send to
an

e-mail address.
Are you aware of a linux based application who does something like this
and

can be installed on the same computer as Asterisk?
Another possibility is a Windows based software SIP fax which can be
registered with Asterisk.
Thanks,
Dan
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Re: [Asterisk-Users] Accurate Billing

[marrandy]

On Sunday 06 July 2003 11:07 pm, [EMAIL PROTECTED] wrote:
> hi everyone,
> I know this issue has been raised many times before, i think still the 
problem remains. When a call is made through a Zap channel, whether it is 
actually made or not (irrespective of whether, engaged, busy, or actually 
answered), asterisk logs it in CDRs as a call made. This makes it impossible 
to do an accurate billing. Has anybody found a way to overcome this problem, 
if yes, please let me/us know.
> Thank you inadvance,
> Surajee
> 

Do you know your sending in HTML ???

-- 
Claret is the liquor for boys; port for men; but he who aspires to be a hero
... must drink brandy.
-- Samuel Johnson

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Re: [Asterisk-Users] Digital phones

[marrandy]

On Sunday 06 July 2003 10:19 pm, Anthony Wood wrote:

> > So does that just leave regular single line phones ?
> > 
> > Besides IP phones.
> > 
> > What else can be used ?
> 
> You can plug your old PABX into asterisk and use its phones through that.
> 
> Softphones (software + computer + soundcard + microphone + speaker)
> Some IP phones have bluetooth gateways so you can use your snazzy mobile 
phone
> without paying your telco.
> 
> cheers,
> Woody


I don't have an old PBX.  But I have access to a supply of 'old'  
business/digital phones if I want them.

Looks like softphones is the way to go


-- 
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[Asterisk-Users] Remote * Using IAX

[Stefano Finetti]

I need to configure an * box to connect to a primary * box which is attached
to a PRI in order to make calls using both the same E1 connection.

I know the best solution is to use IAX (altough i could connect the IP
phones on the remote site directly to the main * box that is VPNed with the
remote one).

I've some doubt on how to config IAX to work in this situation.

I think that:

On the main * box I'll have to configure all the users of the remote site as
IAX users (phone1, phone2, ecc.).

On the remote * i need to set up a "register =>" statement in order to have
the remote * connected to the first one.

Is this right?

In other words, the remote should register on the main * box, but I'm not
sure of where i've to put the phones (all SIP GrandStream phones) to be
registered. In the remote or in the main * IAX.conf file? (or both, maybe).

Any help would be appreciated, since I've not been able to understand on my
own even reading the whole lot of messages about IAX on this list :(

-- 
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
Fax: 06 233 227 934
Linux Registered User #271978

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