[Asterisk-Users] Fax detection on IAX2 channel?

[Guan Yang]

Hi,

Will fax detection work on an IAX2 channel, or is it specific to Zaptel?

Guan

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Re: [Asterisk-Users] Caller-ID for alphanumeric SIP uris

[John Todd]

At 3:02 PM -0700 on 5/11/04, David Beckemeyer wrote:
My first post here, so a brief intro:

I'm somewhat new to Asterisk, but have been working with SIP
in depth for about 3 years.  I studied Asterisk for about a year
and have been experimenting with it hands-on for the past
month or so.  I've done 6 Asterisk installs in wildly different
roles/applications, some of them test systems, others in
semi-production, so I know a little bit about it.  I've setup
voicemail, meetme, ENUM, and other Asterisk features, and I've
written some AGI scripts and done some other semi-interesting
tweaks.
That said, I'm curious about how others might solve the following
problem.  In a pure-SIP environment, if a user has an alphanumeric
SIP uri, say sip:[EMAIL PROTECTED], when that user calls another
SIP phone, (a real IP phone, as opposed to an ATA), via a SIP proxy,
that phone can log the full URI, and 'call return' works because the
SIP phone calls that URI.  With Asterisk, such a call would come in
with the SIP From: header (thus Caller-ID in Asterisk parlance) as
something like:
  From: "joe" ;tag=as54f3792a

In this case, Asterisk doesn't know how to return the call, nor
does the SIP phone, because even if the SIP phone can dial full
alphanumeric URI's with some kind of a 'call return' feature,
the sip:[EMAIL PROTECTED] (where 204.16.112.70 would be the
IP address of the Asterisk server), isn't a valid URI and doesn't
route a call to the original SIP URI: sip:[EMAIL PROTECTED]
I've thought of some tricks for handling this, and I've looked
around the archives and Google searches, but haven't seen much
discussion of this issue.
TIA,

David
David -
  You're correct.  This is an unfortunate side effect of Asterisk not 
really being a SIP proxy.  It's a PBX replacement.  Now, I understand 
that Olle's chan_sip2 has some of this type of feature functionality 
built into it, and you may want to take a look at that.

  In the interim, there are some really awful, terrible, horrendous 
tricks you can do that might work around this problem.  It involves 
snagging the SIP URI on the inbound call, pushing it into a database, 
assigning a pseudo-random number to that entry, and then keeping that 
mapping... forever.  If a user hit the "redial" button, then the 
inverse would happen: your Asterisk server would dig through the 
database looking for the key, find the 'real' SIP URI, and re-route 
the call to the appropriate correct endpoint.  Ugly.

JT
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Re: [Asterisk-Users] Asterisk + VoiceWorks

[Andy Rosen]

What sort of Time Clock?  For employees to punch in and out?  That's a very
attractive idea.  Is this a custom solution or is there a plugin out there
for this?

atr
- Original Message - 
From: "AstGrp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 11, 2004 10:07 PM
Subject: RE: [Asterisk-Users] Asterisk + VoiceWorks


I have a customer who has a Comdial Phone System and uses VoiceWorks for
it's voicemail.  I am installing an IVR / Time Clock system utilizing
asterisk.  But they want to have an option to hop over to the VoiceWorks
system to check voicemail...

-gcc


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brian
Posted At: Tuesday, May 11, 2004 2:52 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Asterisk + VoiceWorks
Subject: RE: [Asterisk-Users] Asterisk + VoiceWorks


Why on earth would you wanna do something like that?  Asterisk has
voicemail and you even have the src so you can add those nifty features
the PHB's like to have but never use!

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of AstGrp
> Sent: Tuesday, May 11, 2004 11:43 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk + VoiceWorks
>
> I have a need to interface Asterisk with a VoiceWorks voicemail
> system. I was wondering what kind of card would be needed either a FXO

> or FXS interface?
>
> Any help would be appreciated.
>
> Thanks,
>
> -gcc
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RE: [Asterisk-Users] Asterisk + VoiceWorks

[AstGrp]

I have a customer who has a Comdial Phone System and uses VoiceWorks for
it's voicemail.  I am installing an IVR / Time Clock system utilizing
asterisk.  But they want to have an option to hop over to the VoiceWorks
system to check voicemail...

-gcc


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brian
Posted At: Tuesday, May 11, 2004 2:52 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Asterisk + VoiceWorks
Subject: RE: [Asterisk-Users] Asterisk + VoiceWorks


Why on earth would you wanna do something like that?  Asterisk has
voicemail and you even have the src so you can add those nifty features
the PHB's like to have but never use!

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of AstGrp
> Sent: Tuesday, May 11, 2004 11:43 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk + VoiceWorks
>
> I have a need to interface Asterisk with a VoiceWorks voicemail 
> system. I was wondering what kind of card would be needed either a FXO

> or FXS interface?
>
> Any help would be appreciated.
>
> Thanks,
>
> -gcc
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Re: [Asterisk-Users] Terrible TICKING sound

[Steven Critchfield]

On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote:

> Our problem ended up not being with Asterisk or Digium hardware.  It was
> the analog cordless phone.  We simply have to live with it.  What
> happens is whenever a connection is established and the phone is
> off-hook, an LED on the base lights up in a "blink blink . blink
> blink . etc." pattern.  Everytime the LED lights, a pulse is sent to
> the phone.  It's especially bad when both lines are in use, as the phone
> is a two-line capable device.  Then you've got double the pulsing.
> 
> This may have nothing to do with your problem.  Just wanted to get it
> out there in case anyone else runs into it, too.

Sounds like your phone needs either a aux power source to power that
led, or possible a little modification to clip that LED.

I would make sure your cordless phone's power supply is within spec. If
it is, Maybe you might want to look into one of the other comments a
while back on the list about upping the power on the SLIC(?). You might
be able to provide enough power to the phone to not cause trouble when
it blinks the LED. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Terrible TICKING sound - Fixed

[Paul Mahler]

Well, for me it was a problem with the T1 line. XO fixed the line and the
ticking sound is gone! 



Paul Mahler 
[EMAIL PROTECTED]   
Signate, LLC
PO Box 60430
Palo Alto, CA
 94306

 VoIP Systems Training & Consulting


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Tuesday, May 11, 2004 3:29 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Terrible TICKING sound

> The Adit channel bank we are using, and XO communications who 
> provisioned the T1 are both showing a LOT of framing errors on our system.

Tell Asterisk to clock from XO's T1.  How is this related to your TDM400P
though?

Regards,
Andrew
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RE: [Asterisk-Users] how to record all agent calls

[Paul Mahler]

you need to combine both sides of the conversation. This 
should be covered in the archives.
 
Paul
 





  
  
Paul 
  Mahler [EMAIL PROTECTED] 
  

  Signate, LLCPO Box 
  60430Palo Alto, CA 94306 VoIP Systems, Training & 
  Consulting
 
 
 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff 
CrewsSent: Tuesday, May 11, 2004 2:57 PMTo: 
[EMAIL PROTECTED]Subject: [Asterisk-Users] how to 
record all agent calls
I want to record incoming calls that are queued when the 
call is connected to an agent.I added the following lines to agents.conf 
before the list of agents:; Enable recording calls addressed to agents. 
It's turned off by default.recordagentcalls=yes;;The format to 
be used to record the calls;wav, gsm, wav49.; By default its 
"wav".recordformat=gsm;; Insert into CDR userfield a name of the the 
created recording; By default it's turned off.createlink=no;; 
The text to be added to the name of the recording. Allows forming a url 
link.;urlprefix=http://host.domain/calls/;; The optional directory 
to save the conversations in. The default is; 
/var/spool/asterisk/monitor;savecallsin=/var/callsand added to the 
queues.conf file:; monitor-format = gsm|wav|wav49monitor-format = 
gsm...and then issued the reload command in the Asterisk CLI 
console.I even created the /var/log/asterisk/monitor directory because 
it did not exist.Is there something else that needs to happen to record 
calls between agents and callers so you can hear both sides of the 
conversation?Thanks in advance.
---Jeff CrewsEastern Oregon Net, Inc.La Grande 
OregonEmail [EMAIL PROTECTED]Voice 541-963-2625 or 800-785-7873,  
extension 11 personal efax 503-907-6704, standard company fax 541-962-7818 
web http://home.eoni.com    

<>

Re: [Asterisk-Users] how to record all agent calls

[brian k. west]

Only works in CVS-HEAD
 
bkw

  - Original Message - 
  From: 
  Jeff Crews 
  To: [EMAIL PROTECTED] 
  
  Sent: Tuesday, May 11, 2004 4:57 PM
  Subject: [Asterisk-Users] how to record 
  all agent calls
  I want to record incoming calls that are queued 
  when the call is connected to an agent.I added the following lines to 
  agents.conf before the list of agents:; Enable recording calls 
  addressed to agents. It's turned off by 
  default.recordagentcalls=yes;;The format to be used to record 
  the calls;wav, gsm, wav49.; By default its 
  "wav".recordformat=gsm;; Insert into CDR userfield a name of the 
  the created recording; By default it's turned 
  off.createlink=no;; The text to be added to the name of the 
  recording. Allows forming a url 
  link.;urlprefix=http://host.domain/calls/;; The optional directory 
  to save the conversations in. The default is; 
  /var/spool/asterisk/monitor;savecallsin=/var/callsand added to the 
  queues.conf file:; monitor-format = gsm|wav|wav49monitor-format = 
  gsm...and then issued the reload command in the Asterisk CLI 
  console.I even created the /var/log/asterisk/monitor directory because 
  it did not exist.Is there something else that needs to happen to 
  record calls between agents and callers so you can hear both sides of the 
  conversation?Thanks in advance.
  ---Jeff CrewsEastern Oregon Net, Inc.La Grande 
  OregonEmail [EMAIL PROTECTED]Voice 541-963-2625 or 800-785-7873,  
  extension 11 personal efax 503-907-6704, standard company fax 541-962-7818 
  web http://home.eoni.com    
  

Re: [Asterisk-Users] Terrible TICKING sound

[Andrew Kohlsmith]

> The Adit channel bank we are using, and XO communications who provisioned
> the T1 are both showing a LOT of framing errors on our system.

Tell Asterisk to clock from XO's T1.  How is this related to your TDM400P 
though?

Regards,
Andrew
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RE: [Asterisk-Users] SDP messages relating to rtpmap Question

[Brian West]

http://lists.cs.columbia.edu/pipermail/sip-implementors/2002-May/003059.html

Thats this same exact issue explained a bit more.

bkw

-- Forwarded message --
Date: Tue, 11 May 2004 17:35:33 -0500
From: brian <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SDP messages relating to rtpmap Question

Actually this has come up before we can receive on one and send on another.
The other end should use what we say.  One leg could be using 103 and the
other leg of the call could use 101.  Or that's how I understand it.

Bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh
> Sent: Tuesday, May 11, 2004 5:17 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] SDP messages relating to rtpmap Question
>
> SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it
> shouldn't
> it send out the same  "a=rtpmap:103 telephone-event/8000"  to the other
> side
> of the connection?  and not  something like "a=rtpmap:101
> telephone-event/8000"?
>
> Thanks
>
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RE: [Asterisk-Users] Terrible TICKING sound

[Paul Mahler]

The Adit channel bank we are using, and XO communications who provisioned
the T1 are both showing a LOT of framing errors on our system.  



Paul Mahler 
[EMAIL PROTECTED]   
Signate, LLC
PO Box 60430
Palo Alto, CA
 94306

 VoIP Systems, Training & Consulting

 

 

 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tmpm
Sent: Tuesday, May 11, 2004 12:14 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Terrible TICKING sound

Ive found this in audio apps on other boxes when the power supply is really
loaded down hard.
Just one more "maybe" for you to check. Have you blown the dust out of the
P/S lately? Dirt and temp variations seem to affect it as well...found this
with audio equipment at a broadcast station. (Streaming server on a Linux
box) Not saying its your situation, but wont hurt to check. If it does help,
a beefier power supply might help here. It cured my case.
Marc

At 10:45 5/11/2004, you wrote:
>I've fought with this problem on and off.
>
>Number 1 thing to check is /proc/interrupts to ensure that your card 
>isn't sharing an interrupt with something else.
>
>Number 2 is a bit of an unknown variable - my guess is either 
>electrical noise, or perhaps vibrations affecting your card inside your 
>box.  I find that carefully remounting my tdm400p/x100p so that nothing 
>at all is touching it (no wires, no plastic, nothing - except at the 
>mount point) will make the problem go away the majority of the time.
>If it doesn't go away, try re-mounting again.
>
>It's a little scary, especially when you're working with a 
>small-form-factor machine.
>Ryan
>
>On 10-May-04, at 8:43 PM, Paul Mahler wrote:
>
>>i'm getting a tick every second or so on all my calls. All channels 
>>are zap channels.
>>
>>Does anyone know how to fix this?
>>
>>Thanks!
>>
>>Paul
>>
>>
>>Paul Mahler
>>[EMAIL PROTECTED]  
>> Signate, LLC PO Box 60430 Palo Alto, CA
>>  94306
>>
>>  VoIP Systems, Training & Consulting
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
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[Asterisk-Users] how to record all agent calls

[Jeff Crews]

I want to record incoming calls that are queued when the
call is connected to an agent.
I added the following lines to agents.conf before the list of
agents:
; Enable recording calls addressed to agents. It's turned off by
default.
recordagentcalls=yes
;
;The format to be used to record the calls
;wav, gsm, wav49.
; By default its "wav".
recordformat=gsm
;
; Insert into CDR userfield a name of the the created recording
; By default it's turned off.
createlink=no
;
; The text to be added to the name of the recording. Allows forming a url
link.
;urlprefix=http://host.domain/calls/
;
; The optional directory to save the conversations in. The default
is
; /var/spool/asterisk/monitor
;savecallsin=/var/calls
and added to the queues.conf file:
; monitor-format = gsm|wav|wav49
monitor-format = gsm
...and then issued the reload command in the Asterisk CLI
console.
I even created the /var/log/asterisk/monitor directory because it did not
exist.
Is there something else that needs to happen to record calls between
agents and callers so you can hear both sides of the
conversation?
Thanks in advance.

---
Jeff Crews
Eastern Oregon Net, Inc.
La Grande Oregon
Email [EMAIL PROTECTED]
Voice 541-963-2625 or 800-785-7873,  extension 11 
personal efax 503-907-6704, standard company fax 541-962-7818 
web
http://home.eoni.com   

RE: [Asterisk-Users] SDP messages relating to rtpmap Question

[brian]

Actually this has come up before we can receive on one and send on another.
The other end should use what we say.  One leg could be using 103 and the
other leg of the call could use 101.  Or that's how I understand it.

Bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh
> Sent: Tuesday, May 11, 2004 5:17 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] SDP messages relating to rtpmap Question
>
> SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it
> shouldn't
> it send out the same  "a=rtpmap:103 telephone-event/8000"  to the other
> side
> of the connection?  and not  something like "a=rtpmap:101
> telephone-event/8000"?
>
> Thanks
>
> ___
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Re: [Asterisk-Users] Terrible TICKING sound

[Jason A. Pattie]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ryan Courtnage wrote:
|
| On 11-May-04, at 8:45 AM, Ryan Courtnage wrote:
|
|> I've fought with this problem on and off.
|>
|> Number 1 thing to check is /proc/interrupts to ensure that your card
|> isn't sharing an interrupt with something else.
|>
|> Number 2 is a bit of an unknown variable - my guess is either
|> electrical noise, or perhaps vibrations affecting your card inside
|> your box.  I find that carefully remounting my tdm400p/x100p so that
|> nothing at all is touching it (no wires, no plastic, nothing - except
|> at the mount point) will make the problem go away the majority of the
|> time.  If it doesn't go away, try re-mounting again.
|
|
| Currently, I'm working with a tdm400p with 2 FXO, 2 FXS.  the persistent
| "tick tick tick tick" has come back (after 3 days tick-free operation).
|
| Has anyone else experienced a similar problem with ZAP channels?  What
| steps did you take to resolve it?
Our problem ended up not being with Asterisk or Digium hardware.  It was
the analog cordless phone.  We simply have to live with it.  What
happens is whenever a connection is established and the phone is
off-hook, an LED on the base lights up in a "blink blink . blink
blink . etc." pattern.  Everytime the LED lights, a pulse is sent to
the phone.  It's especially bad when both lines are in use, as the phone
is a two-line capable device.  Then you've got double the pulsing.
This may have nothing to do with your problem.  Just wanted to get it
out there in case anyone else runs into it, too.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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[Asterisk-Users] SDP messages relating to rtpmap Question

[Glenn Dalgliesh]

SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it shouldn't
it send out the same  "a=rtpmap:103 telephone-event/8000"  to the other side
of the connection?  and not  something like "a=rtpmap:101
telephone-event/8000"?

Thanks

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[Asterisk-Users] Caller-ID for alphanumeric SIP uris

[David Beckemeyer]

My first post here, so a brief intro:

I'm somewhat new to Asterisk, but have been working with SIP
in depth for about 3 years.  I studied Asterisk for about a year 
and have been experimenting with it hands-on for the past 
month or so.  I've done 6 Asterisk installs in wildly different
roles/applications, some of them test systems, others in 
semi-production, so I know a little bit about it.  I've setup
voicemail, meetme, ENUM, and other Asterisk features, and I've 
written some AGI scripts and done some other semi-interesting 
tweaks.

That said, I'm curious about how others might solve the following
problem.  In a pure-SIP environment, if a user has an alphanumeric
SIP uri, say sip:[EMAIL PROTECTED], when that user calls another 
SIP phone, (a real IP phone, as opposed to an ATA), via a SIP proxy,
that phone can log the full URI, and 'call return' works because the
SIP phone calls that URI.  With Asterisk, such a call would come in 
with the SIP From: header (thus Caller-ID in Asterisk parlance) as
something like:

  From: "joe" ;tag=as54f3792a

In this case, Asterisk doesn't know how to return the call, nor
does the SIP phone, because even if the SIP phone can dial full
alphanumeric URI's with some kind of a 'call return' feature, 
the sip:[EMAIL PROTECTED] (where 204.16.112.70 would be the
IP address of the Asterisk server), isn't a valid URI and doesn't
route a call to the original SIP URI: sip:[EMAIL PROTECTED]

I've thought of some tricks for handling this, and I've looked
around the archives and Google searches, but haven't seen much
discussion of this issue.

TIA,

David

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Re: [Asterisk-Users] I love you!

[Soren Rathje]

Nah.. The only way I ever notice that someone sent me a virus is when my mailserver 
says "beep".. And that is currently so frequent that I'm considering turning that 
off.. :-)

Makes me think of this (The Network Auralizer): http://peep.sourceforge.net/intro.html 
Check out the "Low load average demo" mp3, you'll know what I mean..

-- Soren
- Original Message - 
From: "Thomas Gallaway" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, May 10, 2004 9:00 PM
Subject: Re: [Asterisk-Users] I love you!


> [EMAIL PROTECTED] wrote:
> 
> >lovely, :-)
> >  
> >
> Is it just me or where there allready 3 virus sent to this list today?
> Maybe time for denim to disallow attachments? :-)
> 
> -- Thomas
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[Asterisk-Users] midget packet received

[Brian J. Schrock]

Hello,

I just upgraded to the latest cvs and I am seeing these messages in my logs 
now. They are almost always 2 minutes apart, is this a problem? I checked the 
source, appears to be saying the network is not sending the right amount of 
data (recvfrom) when compared to the iax2 mini struct. Seems really odd that 
it happens every 2 minutes, and only after I upgraded to the newest cvs. 
Network interface does not show incrementing errors, carrier transitions etc. 
Sound like a problem with the source not the network, can anyone here set me 
straight?

May 11 16:06:27 WARNING[81926]: File chan_iax2.c, Line 3929 (socket_read): 
midget packet received (0 of 4 min)
May 11 16:08:28 WARNING[81926]: File chan_iax2.c, Line 3929 (socket_read): 
midget packet received (0 of 4 min)
May 11 16:10:28 WARNING[81926]: File chan_iax2.c, Line 3929 (socket_read): 
midget packet received (0 of 4 min)

-- 
Brian J. Schrock
Anistone Technologies, LLC
Phone: 614-798-9106 ext. 2
Cell: 614-537-2817
E-Mail: [EMAIL PROTECTED]
http://www.anistonetech.com
http://www.vipcomtel.com
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Re: [Asterisk-Users] Terrible TICKING sound

[tmpm]

Ive found this in audio apps on other boxes when the power supply is really 
loaded down hard.
Just one more "maybe" for you to check. Have you blown the dust out of the 
P/S lately? Dirt and temp variations seem to affect it as well...found this 
with audio equipment at a broadcast station. (Streaming server on a Linux 
box) Not saying its your situation, but wont hurt to check. If it does 
help, a beefier power supply might help here. It cured my case.
Marc

At 10:45 5/11/2004, you wrote:
I've fought with this problem on and off.

Number 1 thing to check is /proc/interrupts to ensure that your card isn't 
sharing an interrupt with something else.

Number 2 is a bit of an unknown variable - my guess is either electrical 
noise, or perhaps vibrations affecting your card inside your box.  I find 
that carefully remounting my tdm400p/x100p so that nothing at all is 
touching it (no wires, no plastic, nothing - except at the mount point) 
will make the problem go away the majority of the time.
If it doesn't go away, try re-mounting again.

It's a little scary, especially when you're working with a 
small-form-factor machine.
Ryan

On 10-May-04, at 8:43 PM, Paul Mahler wrote:

i'm getting a tick every second or so on all my calls. All channels are zap
channels.
Does anyone know how to fix this?

Thanks!

Paul

Paul Mahler
[EMAIL PROTECTED] 

Signate, LLC
PO Box 60430
Palo Alto, CA
 94306
 VoIP Systems, Training & Consulting









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[Asterisk-Users] Disabling agent call logging

[Jason A. Pattie]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Anyone know how to stop agent calls from being logged?  It appears that
the options in /etc/asterisk/agents.conf do not work with respect to
call logging.  All calls are logged in
/var/spool/asterisk/monitor/.wav, even
with the option to log disabled.  It records 3 files for each call: an
incoming file, an outgoing file, and another file, which I assume is the
merging of incoming and outgoing data (haven't actually listened to any
of these files, yet).
The only reason I'm complaining is that it ate up all the hard drive
space on my first Asterisk system before I realized what was happening
(2.5GB free space ... gone).  I didn't see a need to keep the files, so
I've just been deleting them, but I'd like to not have them recorded in
the first place (or is this something that isn't legal?).
Thanks.

- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAoTJnuYsUrHkpYtARAgILAJ9/emsvzLP4Q3sHCBg4s8K9xh/ltgCgg3bQ
qq9LYjTi+otsVch7AcVSV00=
=W8vW
-END PGP SIGNATURE-
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
MailScanner thanks transtec Computers for their support.
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[Asterisk-Users] Areski CDR graph incorrect

[Gelson Dias Santos]

	Is anyboby using the areski CDR reporting tool?
	I have installed asterisk-stats v1.2 three days ago, but I found a 
possible bug in it. My calls compare graphic shows most of the on the 
calls at first hours past midnight, and it never logs anything after 
lunch time. This is wrong, my calls are made on business hours. The call 
log lists those calls at the right time.
	Is there something I should set on the graphic engine, like a 
timezone or something? I´m on Brazil, timezone GMT -3.
	Tried to email the author directly but got no answer.

Thanks,
Gelson
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Re: [Asterisk-Users] Kernel Freezes with T100P

[Thomas Gallaway]

Zach Chambers wrote:


Ever tried to use the latest fedora kernel. Also what type of chipset 
do you have? Might be an chipset incompatiblity. Did you ever try 
swapping arround the pci card into a nother PCI slot or playing 
arround with the IRQ's?

-- Thomas


Thomas,  I could not get the zaptel drivers to compile using any of 
the fedora kernels which is what drove me to the stock kernel to begin 
with.  I'll check the archives again on that issue.  The chipset is a 
VIA chipset.  I don't know much about it other than that yet.  I'll 
try PCI slot move as well.

Thanks,

-Zach.

I actually just installed the zaptel driver for fedora from cvs.
Christian helped me getting the latest CVS version of asterisk, libpri 
and zaptel.
Then what I did is grabbed the kernel-source for fedora.
I guess in your case you want to do an kernel update.

yum update
yum install kernel-source
It should install the 2188 kernel and 2188 kernel source.
Then just go to /usr/src/zaptel
make clean && make && make install
and same with libpri and asterisk.
Just make sure you removed the wcfxo kernel mod be4 you install zaptel. 
And then
reboot the box as when I modprobe wcfxp it gave me an actuall kernel 
panic :-)
After reboot everything worked just fine. It even seems like my dropped 
call issue
is gone.

If you need a copy of the yum.conf file let me know.

-- Thomas
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RE: [Asterisk-Users] Need help: X100P connection/configuration in GERMANY

[Carlton J. O'Riley]

You need to tell it what to dial. 

exten => 111,1,Dial,Zap/1/


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thorsten Gehrig
Sent: Tuesday, May 11, 2004 2:55 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Need help: X100P connection/configuration in
GERMANY



hi

i need help in connecting a X100P-clone in germany.

 

Basic questions:

a)what configuration do I need ? what is the difference (maybe explain in
german? via email?)

fxs_ls:  FXS (Loop Start)

fxs_gs:  FXS (Ground Start)

fxs_ks:  FXS (Kewl Start)

 

b) can I use one card for connecting to the telephone network (in this case
my analog pbx) and the same card for connecting an analog phone (the card
has two connectors with "line" and "telephone" signs).

 

I want to connect the asterisk pbx to my analog pbx and want to make calls
in booth directions.

with "FXS Krwl Start" I can make calls from the analog-pbx to asterisk (and
connected SIP-phone). but in the other direction "(from asterisk to analog
pbx) I cant make calls. 

the extensions.conf definition is 

exten => 111,1,Dial,Zap/1

but nothing happen if I dial this extension from my sip phone.

 

Executing Dial("SIP/thorstengehrig-5c59", "Zap/1") in new stack

-- Called 1

-- Zap/1-1 answered SIP/thorstengehrig-5c59

 

regards

Thorsten Gehrig (Thorsten at Gehrig.de)

 

 

 

 

 

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Re: [Asterisk-Users] Kernel Freezes with T100P

[Zach Chambers]

Ever tried to use the latest fedora kernel. Also what type of chipset 
do you have? Might be an chipset incompatiblity. Did you ever try 
swapping arround the pci card into a nother PCI slot or playing 
arround with the IRQ's?

-- Thomas
Thomas,  I could not get the zaptel drivers to compile using any of the 
fedora kernels which is what drove me to the stock kernel to begin 
with.  I'll check the archives again on that issue.  The chipset is a 
VIA chipset.  I don't know much about it other than that yet.  I'll try 
PCI slot move as well.

Thanks,

-Zach.

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[Asterisk-Users] Need help: X100P connection/configuration in GERMANY

[Thorsten Gehrig]

hi

i need help in connecting a X100P-clone in germany.

 

Basic questions:

a)what configuration do I need ? what is the difference
(maybe explain in german? via email?)

fxs_ls:  FXS (Loop Start)

fxs_gs:  FXS (Ground Start)

fxs_ks:  FXS (Kewl Start)

 

b) can I use one card for connecting to the telephone
network (in this case my analog pbx) and the same card for connecting an analog
phone (the card has two connectors with “line” and “telephone”
signs).

 

I want to connect the asterisk pbx to my analog pbx
and want to make calls in booth directions.

with “FXS Krwl Start” I can make calls
from the analog-pbx to asterisk (and connected SIP-phone). but in the other
direction “(from asterisk to analog pbx) I cant make calls. 

the extensions.conf definition is 

exten => 111,1,Dial,Zap/1

but nothing happen if I dial this extension from my
sip phone.

 

Executing Dial("SIP/thorstengehrig-5c59",
"Zap/1") in new stack

    -- Called 1

    -- Zap/1-1 answered
SIP/thorstengehrig-5c59

 

regards

Thorsten Gehrig (Thorsten at Gehrig.de)

 

 

 

 

 

Re: [Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting

[Tony]

Take a look here: http://www.voip-info.org/wiki-Asterisk+Avaya 
t o n y
On Tue, 2004-05-11 at 13:38, brian k. west wrote:
> You have to fix the dialplan or it won't work.  I think it hijacks em by
> default.
> 
> bkw
> - Original Message - 
> From: "PBX Tech" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, May 11, 2004 10:22 AM
> Subject: [Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting
> 
> 
> >
> > I have searched through all the lists and found that some people have had
> > luck with flash hook, then *0 to answer a call waiting call.
> >
> > I have an Asterisk server with one FXO card, the dialtone for the fxo card
> > is providing by another pbx called a Definity.
> >
> > When I am on the Sipura, and another call comes in, I hear the
> call-waiting
> > indicator, when I flash hook I just hear tone, if I dial *0 I just hear
> dead
> > air.
> >
> > Its like the Asterisk isnt flashing the fxo line.
> >
> > Any suggestions?
> >
> > _
> > MSN Toolbar provides one-click access to Hotmail from any Web page - FREE
> > download! http://toolbar.msn.com/go/onm00200413ave/direct/01/
> >
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RE: [Asterisk-Users] Asterisk + VoiceWorks

[brian]

Why on earth would you wanna do something like that?  Asterisk has voicemail
and you even have the src so you can add those nifty features the PHB's like
to have but never use!

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of AstGrp
> Sent: Tuesday, May 11, 2004 11:43 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk + VoiceWorks
>
> I have a need to interface Asterisk with a VoiceWorks voicemail system.
> I was wondering what kind of card would be needed either a FXO or FXS
> interface?
>
> Any help would be appreciated.
>
> Thanks,
>
> -gcc
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[Asterisk-Users] KIRK System 600 IP and Asterisk

[Bartosz Jozwiak]

Hello,

Does anybody try to connect KIRK System 600 IP to asterisk ?

http://www.kirktelecom.dk/company/suk207.asp

B.

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Re: [Asterisk-Users] Terrible TICKING sound

[Andrew Kohlsmith]

> Currently, I'm working with a tdm400p with 2 FXO, 2 FXS.  the
> persistent "tick tick tick tick" has come back (after 3 days tick-free
> operation).
>
> Has anyone else experienced a similar problem with ZAP channels?  What
> steps did you take to resolve it?

Are you seeing anything in dmesg?  after you've heard some ticking, run dmesg 
and look for anything suspicious about the TDM card; things about power 
module resets and stuff.  I find that when I hear a good click or buzz I see 
that the power supply farted or otherwise confused the card.

Regards,
Andrew
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Re: [Asterisk-Users] Terrible TICKING sound

[Ryan Courtnage]

On 11-May-04, at 8:45 AM, Ryan Courtnage wrote:

I've fought with this problem on and off.

Number 1 thing to check is /proc/interrupts to ensure that your card 
isn't sharing an interrupt with something else.

Number 2 is a bit of an unknown variable - my guess is either 
electrical noise, or perhaps vibrations affecting your card inside 
your box.  I find that carefully remounting my tdm400p/x100p so that 
nothing at all is touching it (no wires, no plastic, nothing - except 
at the mount point) will make the problem go away the majority of the 
time.  If it doesn't go away, try re-mounting again.
Currently, I'm working with a tdm400p with 2 FXO, 2 FXS.  the 
persistent "tick tick tick tick" has come back (after 3 days tick-free 
operation).

Has anyone else experienced a similar problem with ZAP channels?  What 
steps did you take to resolve it?

Cheers


It's a little scary, especially when you're working with a 
small-form-factor machine.
Ryan

On 10-May-04, at 8:43 PM, Paul Mahler wrote:

i'm getting a tick every second or so on all my calls. All channels 
are zap
channels.

Does anyone know how to fix this?

Thanks!

Paul

Paul Mahler
[EMAIL PROTECTED] 

Signate, LLC
PO Box 60430
Palo Alto, CA
 94306
 VoIP Systems, Training & Consulting

	







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Re: [Asterisk-Users] Kernel Freezes with T100P

[Thomas Gallaway]

Zach Chambers wrote:

Hey Folks,

I have an extremely reproducable problem with the T100P card.  I'm 
running one card in an Athlon XP 1800 machine running Fedora Core 1.  
The kernel is a stock 2.4.26 kernel pulled from kernel.org.  I'm going 
to be using the T1 interface for data only.  If I bring the interface 
up, then down, then up again, the machine will freeze within 30 
seconds or so.  There are are no debugging messages even if debugging 
is turned on.  Occasionally  it takes another up/down cycle to cause 
the machine to freeze, but that is rare.  I've seen other incidents of 
machine's freezing in the archives, but no real answers to the 
problem.  Any idea where to start with this?

Thanks,

-Zach.
Ever tried to use the latest fedora kernel. Also what type of chipset do 
you have? Might be an chipset incompatiblity. Did you ever try swapping 
arround the pci card into a nother PCI slot or playing arround with the 
IRQ's?

-- Thomas
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[Asterisk-Users] Kernel Freezes with T100P

[Zach Chambers]

Hey Folks,

I have an extremely reproducable problem with the T100P card.  I'm 
running one card in an Athlon XP 1800 machine running Fedora Core 1.  
The kernel is a stock 2.4.26 kernel pulled from kernel.org.  I'm going 
to be using the T1 interface for data only.  If I bring the interface 
up, then down, then up again, the machine will freeze within 30 seconds 
or so.  There are are no debugging messages even if debugging is turned 
on.  Occasionally  it takes another up/down cycle to cause the machine 
to freeze, but that is rare.  I've seen other incidents of machine's 
freezing in the archives, but no real answers to the problem.  Any idea 
where to start with this?

Thanks,

-Zach.

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Re: [Asterisk-Users] * INSTRUCTIONS FOR MEMBERS OF THE COMMUNITY * Please read

[Andrew Kohlsmith]

> Might I make the request that the next time you post this, you do NOT
> reply to a message that's two months old?  Those of us with threaded
> mail readers have to page back two months in our histories to read
> your message.

I use KMail's next/prev new message to catch all of that.  :-)

Regards,
Andrew
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RE: [Asterisk-Users] Error compiling Zaptel

[Ben Merrills]

Do you have a symlink in /usr/src as
follows?

 

lrwxrwxrwx    1 root src    20
May  7 11:01 linux -> kernel-source-2.4.18

 

(note that it may differ depending on the
kernel source you have?)

 

If you’ve installed via an apt style
package manager, and havn’t recompiled your kernel, then visit www.kernel.org, download a stable kernel (I recommended
2.4.26).

 

Extract it to
/usr/src/linux-

 

Then create a symlink ln –s /usr/src/ /usr/src/linux

 

This should resolve the issues you’re
having there, else I’ve missed the point and just waffled for 5 minutes
;)

 

Hope that helps,

 

Ben Merrills

Griffin Internet

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of San Singhania
Sent: 11 May 2004 16:17
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Error
compiling Zaptel



 



Hi,





 





I just finished downloading asterisk and when trying to compile
the zaptel drivers, get the following errors. I dont have a clue whats going
on...





can someone help.





 





In file included from /usr/include/linux/module.h:20,
from zaptel.c:44:
/usr/include/linux/modversions.h:1:2:#error Modules should never use
kernel-headers system headers,
/usr/include/linux/modversions.h:2:2: error but rather headers from an
appropriate kernel-source package.
/usr/include/linux/modversions.h:3:2: #error Change -I/usr/src/linux/include
(or similar) to
/usr/include/linux/modversions.h:4:2: #error -I/lib/modules/$(uname
-r)/build/include
/usr/include/linux/modversions.h:5:2: #error to build against the
currently-running kernel.
make: *** [zaptel.o] Error 1





 





Thanks





 





San





 

Re: [Asterisk-Users] What is the most popular pre-paid billing system?

[David Liu]

Hi there,

We are writing one right now from scratch using MySQL as backend.  We
will probably release it open source once the code is mature... It will
work with Asterisk via AGI.

David
Deltapath



>>> [EMAIL PROTECTED] 04/04/04 7:47 PM >>>
 

We have a system with multiple Asterisk gateways and an SER proxy. I was
wondering if people could share what they have been doing for prepaid
billing on either multiple Asterisk gateways, or a single SER proxy?

 

Do people use a radius or do they go right from the mysql database? I
have
seen people talking about both, but I was wondering if anyone has it
working.

 

 


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[Asterisk-Users] Asterisk + VoiceWorks

[AstGrp]

I have a need to interface Asterisk with a VoiceWorks voicemail system.
I was wondering what kind of card would be needed either a FXO or FXS
interface?

Any help would be appreciated.

Thanks,

-gcc
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Re: [Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting

[brian k. west]

You have to fix the dialplan or it won't work.  I think it hijacks em by
default.

bkw
- Original Message - 
From: "PBX Tech" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 11, 2004 10:22 AM
Subject: [Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting


>
> I have searched through all the lists and found that some people have had
> luck with flash hook, then *0 to answer a call waiting call.
>
> I have an Asterisk server with one FXO card, the dialtone for the fxo card
> is providing by another pbx called a Definity.
>
> When I am on the Sipura, and another call comes in, I hear the
call-waiting
> indicator, when I flash hook I just hear tone, if I dial *0 I just hear
dead
> air.
>
> Its like the Asterisk isnt flashing the fxo line.
>
> Any suggestions?
>
> _
> MSN Toolbar provides one-click access to Hotmail from any Web page - FREE
> download! http://toolbar.msn.com/go/onm00200413ave/direct/01/
>
> ___
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RE: [Asterisk-Users] Error compiling Zaptel

[Jay Milk]

Title: Message



Your 
question would be better posed on a Linux site, as you're having problems with 
your linux installation, not so much with Asterisk itself.  Search google 
for "linux recompile kernel" and add the actual linux distro as a search term 
(redhat, debian, freebsd, mandrake, whatever).  See if you find your 
answer, if not, ask in the same place you find close 
matches.

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of San 
  SinghaniaSent: Tuesday, May 11, 2004 11:11 AMTo: 
  [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Error 
  compiling Zaptel
  Ok, how do I go about doing that? I am not exactly a linux expert, infact 
  this is the 
  2nd time I am installing it. I would really appreciate some direction on 
  this.
   
  Thanks
   
  San
   
   
   
  > Lets draw your attention to these excerpts of the errors you sent. 
  >Once you read those error messages, you should formulate the 
  opinion>that having the kernel source for your running kernel is 
  IMPORTANT. Then>you will also need to make sure you have the proper 
  modversions.h built>by the kernel source too. >-- >Steven 
  Critchfield  <[EMAIL PROTECTED]>

[Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting

[PBX Tech]

I have searched through all the lists and found that some people have had 
luck with flash hook, then *0 to answer a call waiting call.

I have an Asterisk server with one FXO card, the dialtone for the fxo card 
is providing by another pbx called a Definity.

When I am on the Sipura, and another call comes in, I hear the call-waiting 
indicator, when I flash hook I just hear tone, if I dial *0 I just hear dead 
air.

Its like the Asterisk isnt flashing the fxo line.

Any suggestions?

_
MSN Toolbar provides one-click access to Hotmail from any Web page – FREE 
download! http://toolbar.msn.com/go/onm00200413ave/direct/01/

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Re: [Asterisk-Users] Error compiling Zaptel

[San Singhania]

Ok, how do I go about doing that? I am not exactly a linux expert, infact 
this is the 
2nd time I am installing it. I would really appreciate some direction on 
this.
 
Thanks
 
San
 
 
 
> Lets draw your attention to these excerpts of the errors you sent. 
>Once you read those error messages, you should formulate the 
opinion>that having the kernel source for your running kernel is 
IMPORTANT. Then>you will also need to make sure you have the proper 
modversions.h built>by the kernel source too. >-- >Steven 
Critchfield  <[EMAIL PROTECTED]>

Re: [Asterisk-Users] * INSTRUCTIONS FOR MEMBERS OF THE COMMUNITY * Please read

[Tilghman Lesher]

On Tuesday 11 May 2004 01:10, Olle E. Johansson wrote:
> References: <[EMAIL PROTECTED]>
> In-Reply-To: <[EMAIL PROTECTED]>
> 
> Welcome to the Asterisk users community!

Might I make the request that the next time you post this, you do NOT
reply to a message that's two months old?  Those of us with threaded
mail readers have to page back two months in our histories to read
your message.

-Tilghman

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Re: [Asterisk-Users] Error compiling Zaptel

[Steven Critchfield]

On Tue, 2004-05-11 at 10:16, San Singhania wrote:
> Hi,
>  
> I just finished downloading asterisk and when trying to compile the
> zaptel drivers, get the following errors. I dont have a clue whats
> going on...
> can someone help.
>  

> /usr/include/linux/modversions.h:1:2:#error Modules should never use
> kernel-headers system headers,
> /usr/include/linux/modversions.h:2:2: error but rather headers from an
> appropriate kernel-source package.

> /usr/include/linux/modversions.h:5:2: #error to build against the
> currently-running kernel.
> make: *** [zaptel.o] Error 1

Lets draw your attention to these excerpts of the errors you sent. 

Once you read those error messages, you should formulate the opinion
that having the kernel source for your running kernel is IMPORTANT. Then
you will also need to make sure you have the proper modversions.h built
by the kernel source too. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] Error compiling Zaptel

[San Singhania]

Hi,
 
I just finished downloading asterisk and when trying to compile the 
zaptel drivers, get the following errors. I dont have a clue whats going 
on...
can someone help.
 
In file included from /usr/include/linux/module.h:20,from 
zaptel.c:44:/usr/include/linux/modversions.h:1:2:#error Modules should never 
use kernel-headers system headers,/usr/include/linux/modversions.h:2:2: 
error but rather headers from an appropriate kernel-source 
package./usr/include/linux/modversions.h:3:2: #error Change 
-I/usr/src/linux/include (or similar) 
to/usr/include/linux/modversions.h:4:2: #error -I/lib/modules/$(uname 
-r)/build/include/usr/include/linux/modversions.h:5:2: #error to build 
against the currently-running kernel.make: *** [zaptel.o] Error 1
 
Thanks
 
San
 

RE: [Asterisk-Users] Line appearances

[Adams, Gavin]

With the same extension presented twice (e.g., 2000), is there anyway to
have the phone not ring on the 2nd presentation of the line if first
presentation is currently engaged?

Regards,

--- Gavin
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[Asterisk-Users] New patch for Bug 1420

[Frank Mandarino]

Hi,

I just wanted to point out that I have added a patch to Bug #1420 
(http://bugs.digium.com/bug_view_page.php?bug_id=0001420), which adds 
Manager Events and Queue Log records for the AgentCallbackLogin application.

Since the Bug Summary starts with "[Request]" and the Severity is 
"Trivial", both of which I can't change, this patch might be easy to 
overlook.

Would it make sense to open a new Bug for this patch, or can I impose 
upon someone with more privilege to update the Summary line?

Thanks,
../fam
--
Frank A. Mandarino   [EMAIL PROTECTED]
Spindrift Management, Toronto
416 642-3404
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Re: [Asterisk-Users] Line appearances

[Brian Cuthie]

That's exactly what it means. And, Asterisk will do the right thing.

-brian

Joseph wrote:

Thanks for the help.

Does that mean that say I had an extension 240 on
line appearance one, I could make button 2,3,4 also register
the same ext number?
Would * care that there were multiple entries from the same ip
for the same ext?
Thanks again for the tips.

On Tue, 2004-05-11 at 09:28, Brian Cuthie wrote:
 

Hi Joseph,

I'll assume you mean a 7960 with the SIP image...

Yes, you can register the same SIP client multiple times. Each line 
appearance on a 7960 configured for SIP is a separate SIP client. Each 
can register with completely different SIP proxies (providers) or you 
can have several registrations for the same directory number (DN) so 
that instead of call waiting, additional calls appear at the next 
available line appearance.

I can't answer your coded question since I always use g.711ulaw.

-brian

Joseph wrote:

   

I am trying to get an understanding of how line appearances work
like on the cisco 7960 phones.
Is there a wiki somewhere about how this works?

Also, the 7960 phones let you register more than one ext.
Why would you want more than one or is this connected to line
appearances?
Is there a way to have phones use more than one codec, say
use g.711 to talk with * and g.729 to talk with another
phone?


 

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[Asterisk-Users] Long delays when talking betwwen SIP phones

[Greg Scasny]

I am experiencing some long delays when talking SIP to SIP (I
will talk, it it takes about 2 seconds for that
conversation to reach the other SIP phone, and when they talk back, same thing)
on our local LAN. Traffic is relatively light on our LAN. But talking SIP to
PSTN works fine, no delays. If anyone has experienced similar delays, what have
you done to correct it?

 

I have tried different codecs (GSM
to GSM, ulaw to ulaw) and
still the same result.

 

I truly think this that the problem lies within the LAN, but
wanted to make sure that others have not had the same problem and it turned out
to be a config issue.

 

Any assistance is greatly appreciated.

 

Thanks in advance.>>Greg

 

Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200

 

Re: [Asterisk-Users] Terrible TICKING sound

[Ryan Courtnage]

I've fought with this problem on and off.

Number 1 thing to check is /proc/interrupts to ensure that your card 
isn't sharing an interrupt with something else.

Number 2 is a bit of an unknown variable - my guess is either 
electrical noise, or perhaps vibrations affecting your card inside your 
box.  I find that carefully remounting my tdm400p/x100p so that nothing 
at all is touching it (no wires, no plastic, nothing - except at the 
mount point) will make the problem go away the majority of the time.  
If it doesn't go away, try re-mounting again.

It's a little scary, especially when you're working with a 
small-form-factor machine.
Ryan

On 10-May-04, at 8:43 PM, Paul Mahler wrote:

i'm getting a tick every second or so on all my calls. All channels 
are zap
channels.

Does anyone know how to fix this?

Thanks!

Paul

Paul Mahler
[EMAIL PROTECTED] 

Signate, LLC
PO Box 60430
Palo Alto, CA
 94306
 VoIP Systems, Training & Consulting

	







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Re: [Asterisk-Users] help-listening to my mailbox

[Steve Totaro]

add this to extensions.conf

exten => 55,1,voicemailMain  (replace 55 with whatever you want)
exten => 55,2,Hangup

reload and dial 55


- Original Message - 
From: "leonardo" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, May 10, 2004 7:08 PM
Subject: [Asterisk-Users] help-listening to my mailbox


> How can I listen to the masseges that I have in my mailbox?
> 
> regards
> 
> leo
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Re: [Asterisk-Users] Line appearances

[Fran Boon]

Joseph wrote:
I am trying to get an understanding of how line appearances work
like on the cisco 7960 phones.
Is there a wiki somewhere about how this works?
General info on 79XX:
http://voip-info.org/wiki-Asterisk+phone+cisco+79xx
Also, the 7960 phones let you register more than one ext.
Why would you want more than one or is this connected to line
appearances?
Yes

Is there a way to have phones use more than one codec, say
use g.711 to talk with * and g.729 to talk with another
phone?
No

F
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[Asterisk-Users] help-listening to my mailbox

[leonardo]

How can I listen to the masseges that I have in my mailbox?

regards

leo
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Re: [Asterisk-Users] Line appearances

[Joseph]

Thanks for the help.

Does that mean that say I had an extension 240 on
line appearance one, I could make button 2,3,4 also register
the same ext number?
Would * care that there were multiple entries from the same ip
for the same ext?

Thanks again for the tips.

On Tue, 2004-05-11 at 09:28, Brian Cuthie wrote:
> Hi Joseph,
> 
> I'll assume you mean a 7960 with the SIP image...
> 
> Yes, you can register the same SIP client multiple times. Each line 
> appearance on a 7960 configured for SIP is a separate SIP client. Each 
> can register with completely different SIP proxies (providers) or you 
> can have several registrations for the same directory number (DN) so 
> that instead of call waiting, additional calls appear at the next 
> available line appearance.
> 
> I can't answer your coded question since I always use g.711ulaw.
> 
> -brian
> 
> Joseph wrote:
> 
> >I am trying to get an understanding of how line appearances work
> >like on the cisco 7960 phones.
> >
> >Is there a wiki somewhere about how this works?
> >
> >Also, the 7960 phones let you register more than one ext.
> >Why would you want more than one or is this connected to line
> >appearances?
> >
> >Is there a way to have phones use more than one codec, say
> >use g.711 to talk with * and g.729 to talk with another
> >phone?
> >
> >  
> >
> 
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-- 
respectfully, Joseph - (606) 477-2355 x140
   --=

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Re: [Asterisk-Users] Line appearances

[Brian Cuthie]

Hi Joseph,

I'll assume you mean a 7960 with the SIP image...

Yes, you can register the same SIP client multiple times. Each line 
appearance on a 7960 configured for SIP is a separate SIP client. Each 
can register with completely different SIP proxies (providers) or you 
can have several registrations for the same directory number (DN) so 
that instead of call waiting, additional calls appear at the next 
available line appearance.

I can't answer your coded question since I always use g.711ulaw.

-brian

Joseph wrote:

I am trying to get an understanding of how line appearances work
like on the cisco 7960 phones.
Is there a wiki somewhere about how this works?

Also, the 7960 phones let you register more than one ext.
Why would you want more than one or is this connected to line
appearances?
Is there a way to have phones use more than one codec, say
use g.711 to talk with * and g.729 to talk with another
phone?
 

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Re: [Asterisk-Users] AGI.pm wait_for_digit() not working for me!!!

[Andy Powell]

Ok,

the first think to do is check the permissions on the conf-background.agi ..asterisk 
needs to be able to run it ...

The code I've listed below works fine for me:

#!/usr/bin/perl

use Asterisk::AGI;

$AGI = new Asterisk::AGI;
%input = $AGI->ReadParse();

$soundpath = "/var/lib/asterisk/sounds/";
$timeout = 10;


while(1)
{
$input = chr($AGI->wait_for_digit($timeout));

if ($input eq "*")
{
$AGI->stream_file("$soundpath/banana-phone-song");
}

if ($input eq "1")
{
exit 0;
}

}




*** REPLY SEPARATOR  ***

On 11/05/2004 at 10:52 Atif wrote:

>Hello everybody!!!
>
>
>
>I really need your help guys, I am using the AGI mode in meetme
>application,
>and  I want that AGI should wait for an input from the client/user i.e. a
>digit and then proceed, but I have used that AGI function
>agi->wait_for_digit(), but no usemy agi just passes, or ignores this
>function,
>
>where AGI should stop here and wait for the input
>
>
>
>.my extension in my dialplan.
>
>exten => 21,1,answer
>
>exten => 21,2,meetme(21|pb)
>
>
>
>..and here is my AGI...
>
>#!/usr/bin/perl -w
>
>#use strict;
>
>
>
>$aginame="conf-background.agi";
>
>use File::Copy cp;
>
>use Asterisk::AGI;
>
>$AGI = new Asterisk::AGI;
>
>my %input = $AGI->ReadParse();
>
>
>
>$char=0;
>
>
>
>#while(1)
>
>{
>
>
>
>#$AGI->exec('WaitExten','25000');
>
>#$char = $AGI->receive_char('600');
>
>$char=chr($AGI->wait_for_digit('600'));
>
>
>
>print STDERR "input form rec char : $char\n";
>
>
>
>if($char eq "*")
>
>{
>
>print STDERR "Dialing your number\n";
>
>$srcfile="/tmp/mycall";
>
>$dstfile="/var/spool/asterisk/outgoing/mycall";
>
>open(MYCALL,">$srcfile") || die "Cant't open file :$srcfile
>$!\n";
>
>print MYCALL "Channel:IAX2/bali:[EMAIL PROTECTED]/[EMAIL PROTECTED]";
>
>print MYCALL "MaxRetries:2\n";
>
>print MYCALL "RetryTime:60\n";
>
>print MYCALL "WaitTime:30\n";
>
>print MYCALL "Context:atif\n";
>
>print MYCALL "Extension:22\n";
>
>print MYCALL "Priority:1\n";
>
>close MYCALL;
>
>#   cp($srcfile,$dstfile);
>
>print STDERR "dialing complete...\n";
>
>}
>
>
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[Asterisk-Users] RE: Sipgate to regular phones

[nicolas]

Thanks

Now it works.

nicolas

Simon Brown wrote:

> I just came across the same problem.  I fixed it by changing from
> sipgate.net to sipgate.de in the sip.conf file.
> 
> Simon
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of nicolas
> Sent: Tuesday, 11 May 2004 21:10
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Sipgate to regular phones
> 
> I could call a regular phone through sipgate.
> Now i can not:
> 
> Failed to authenticate on INVITE to '"xyz"
> ;tag=as4ddd4a6f'
> 
> A call from outside to my sip-phone through sipgate is OK.
> 
> Can anyone verify ?
> Is it a sipgate problem ?
> 
> greetings nicolas
> 
> 
> 
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Re: [Asterisk-Users] Asterisk resource consumption..

[WipeOut]

Robert Lawrence wrote:

Unfortunately, it is not as simple as just counting channel types.  
You need to factor in which compression codecs are used, as each will 
have a different impact on the system.  Are multiple codecs used 
requiring transcoding, or just one?  And what is Asterisk being used 
for?   Is Asterisk monitoring the channels or is native bridging 
permitted?  Is muisc on hold used, and if so, how many different music 
on hold contexts?  Is Asterisk recording the channels?  What about 
meetme conferences?

Regards.


All valid points..

I realise that it would not be a perfect calculation but if it was 
possible to say that in an average you need X MB for running Asterisk 
and then Y MB per active channel it would at least give people some 
method to calculate the memory requirements, this is similar to most 
software where they will say you need a certain amout of memory to run 
the app and then in server systems they tell you to add an amount per user..

For example, my system uses 83 MB of RAM (excluding buffers and cache) 
when idle and when I make a call to the echo test it uses about 20k.. 
This means that by these numbers a standard system for 100 concurrent 
users would need no more than  128MB of RAM minimum and 256MB would be 
plenty.. Obviously this is very simplistic and only based on a echo test 
but it does mean that the people who are throwing 2GB of memory at their 
asterisk servers are wasting a lot of money that could have been put 
into faster processors to handle more calls and services..

Thats really all I was talking about so that it helps people size their 
systems and would probably mean fewer "What system do I need?" questions..

Later..


WipeOut wrote:

No this is not another of the "What hardware do I need?" posts.. :)

Just wondering if anyone has calculated the memory consumprion for 
running asterisk..

For example, when its idle it uses U MB or RAM,  uses V MB for each 
active Zap channel, W MB for each active SIP channel, X MB for each 
active IAX Channel and Y MB for each VM channel..

I often read posts where guys are throwing 1-2GB or RAM at a server 
and this seems extreme to me but there don't seem to be any numbers 
for people to use as a guide (could be placed on voip-info.org)..

Unfortunately my system is to small to do any real testing of this 
kind but I would be happy to help somone who has a bigger more active 
system to get the numbers..

Later..
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RE: [Asterisk-Users] Sipgate to regular phones

[Simon Brown]

I just came across the same problem.  I fixed it by changing from sipgate.net
to sipgate.de in the sip.conf file.

Simon 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nicolas
Sent: Tuesday, 11 May 2004 21:10
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sipgate to regular phones

I could call a regular phone through sipgate.
Now i can not:

Failed to authenticate on INVITE to '"xyz"
;tag=as4ddd4a6f'

A call from outside to my sip-phone through sipgate is OK.

Can anyone verify ?
Is it a sipgate problem ?

greetings nicolas



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Re: [Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!

[Steven Critchfield]

On Sun, 2004-05-09 at 07:41, Mark Elkins wrote:
> On Sun, 2004-05-09 at 14:33, Mark Elkins wrote:
> > On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote:
> > > * Read the config sample files! (even if you're an Asterisk guru)
> > > -
> > > For those of you that have a working installation that you keep using, this is a
> > > reminder to check into the configs/ directory of the Asterisk source tree, 
> > > regardless
> > > if you downloaded a tar ball or from CVS.
> > 
> > Good advice - so I do a CVS UPDATE... and 'say.c' is broken
> ...
> > The lines that begin with "<<< say.c"
> 
> Sorry folks... seems like a CVS Update did break - removed the file and
> re-updated. fine now.
> 
> However - this could bit other people too.. in which case - delete the
> offending file - and update again (or always use 'cvs checkout' - less
> efficient - but..)

There should have been a warning from CVS about not being able to merge
the changes to that files. Then you would have been warned, and you
might have learned then that the sections that had been marked by the
'' would have shown you where you needed to reconsile the
changes to say.c. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Asterisk resource consumption..

[Robert Lawrence]

Unfortunately, it is not as simple as just counting channel types.  You 
need to factor in which compression codecs are used, as each will have a 
different impact on the system.  Are multiple codecs used requiring 
transcoding, or just one?  And what is Asterisk being used for?   Is 
Asterisk monitoring the channels or is native bridging permitted?  Is 
muisc on hold used, and if so, how many different music on hold 
contexts?  Is Asterisk recording the channels?  What about meetme 
conferences?

Regards.



WipeOut wrote:

No this is not another of the "What hardware do I need?" posts.. :)

Just wondering if anyone has calculated the memory consumprion for 
running asterisk..

For example, when its idle it uses U MB or RAM,  uses V MB for each 
active Zap channel, W MB for each active SIP channel, X MB for each 
active IAX Channel and Y MB for each VM channel..

I often read posts where guys are throwing 1-2GB or RAM at a server 
and this seems extreme to me but there don't seem to be any numbers 
for people to use as a guide (could be placed on voip-info.org)..

Unfortunately my system is to small to do any real testing of this 
kind but I would be happy to help somone who has a bigger more active 
system to get the numbers..

Later..
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Re: [Asterisk-Users] Sipgate to regular phones

[markus monka]

> Failed to authenticate on INVITE to '"xyz"
> ;tag=as4ddd4a6f'

what says "sip debug"?

greets
Markus


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[Asterisk-Users] Sipgate to regular phones

[nicolas]

I could call a regular phone through sipgate.
Now i can not:

Failed to authenticate on INVITE to '"xyz"
;tag=as4ddd4a6f'

A call from outside to my sip-phone through sipgate is OK.

Can anyone verify ?
Is it a sipgate problem ?

greetings nicolas



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[Asterisk-Users] Line appearances

[Joseph]

I am trying to get an understanding of how line appearances work
like on the cisco 7960 phones.

Is there a wiki somewhere about how this works?

Also, the 7960 phones let you register more than one ext.
Why would you want more than one or is this connected to line
appearances?

Is there a way to have phones use more than one codec, say
use g.711 to talk with * and g.729 to talk with another
phone?

-- 
respectfully, Joseph


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RE: [Asterisk-Users] Use buttons (other than #) after call is bridged?

[Mickey Binder]

That should be pretty easy if you already have the interface to the DPIN I/O
port. Then it would be just a matter of some system() calls on different
extensions which communicate with your DPIN I/O port program.

e.x.:

exten => 99910,1,system(io_prog 1 0) ;turn port 1 off
exten => 99911,1,system(io_prog 1 1) ;turn port 1 on
exten => 99920,1,system(io_prog 2 0) ;turn port 2 off
exten => 99921,1,system(io_prog 2 1) ;turn port 2 on
exten => 99930,1,system(io_prog 3 0) ;turn port 3 off
exten => 99931,1,system(io_prog 3 1) ;turn port 3 on

I don't know if this is the optimal solution, but I would implement it that
way if is was my project.

Regards,
Mickey

> -Original Message-
> From: Dean Collins [mailto:[EMAIL PROTECTED]
> Sent: 11. maj 2004 12:09
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Use buttons (other than #) after call is
> bridged?
> 
> Is there some way of driving external contacts with Asterisk?
> 
> I've seen something running on windows that allowed a Dpin I/O port to
> drive up to 15 contacts, is there someway to get asterisk to do the
> same?
> 
> 
> Cheers,
> Dean
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Andreas
> Anderson
> Sent: Tuesday, 11 May 2004 7:52 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Use buttons (other than #) after call is
> bridged?
> 
> Hi,
> 
> can i somehow use the other buttons to execute some apps, *without*
> hanging
> up the call?
> Something like:
> 
> exten => s,1,Dial/SIP(1234)|4,5,7,9
> exten => 4,1,Monitor(wav)
> exten => 5,1,SIPDtmfMode(inband)
> exten => 7,1,AGI(turnoncoffeemachine.agi)
> exten => 9,1,System(smbnuke boss)
> 
> 
> Regards,
> 
> AA
> 
> _
> Watch movie trailers online with the Xtra Broadband Channel
> http://xtra.co.nz/broadband
> 
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RE: [Asterisk-Users] Use buttons (other than #) after call is bridged?

[Dean Collins]

Is there some way of driving external contacts with Asterisk?

I've seen something running on windows that allowed a Dpin I/O port to
drive up to 15 contacts, is there someway to get asterisk to do the
same?


Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Anderson
Sent: Tuesday, 11 May 2004 7:52 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Use buttons (other than #) after call is
bridged?

Hi,

can i somehow use the other buttons to execute some apps, *without*
hanging 
up the call?
Something like:

exten => s,1,Dial/SIP(1234)|4,5,7,9
exten => 4,1,Monitor(wav)
exten => 5,1,SIPDtmfMode(inband)
exten => 7,1,AGI(turnoncoffeemachine.agi)
exten => 9,1,System(smbnuke boss)


Regards,

AA

_
Watch movie trailers online with the Xtra Broadband Channel  
http://xtra.co.nz/broadband

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[Asterisk-Users] Asterisk resource consumption..

[WipeOut]

No this is not another of the "What hardware do I need?" posts.. :)

Just wondering if anyone has calculated the memory consumprion for 
running asterisk..

For example, when its idle it uses U MB or RAM,  uses V MB for each 
active Zap channel, W MB for each active SIP channel, X MB for each 
active IAX Channel and Y MB for each VM channel..

I often read posts where guys are throwing 1-2GB or RAM at a server and 
this seems extreme to me but there don't seem to be any numbers for 
people to use as a guide (could be placed on voip-info.org)..

Unfortunately my system is to small to do any real testing of this kind 
but I would be happy to help somone who has a bigger more active system 
to get the numbers..

Later..
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[Asterisk-Users] Use buttons (other than #) after call is bridged?

[Andreas Anderson]

Hi,

can i somehow use the other buttons to execute some apps, *without* hanging 
up the call?
Something like:

exten => s,1,Dial/SIP(1234)|4,5,7,9
exten => 4,1,Monitor(wav)
exten => 5,1,SIPDtmfMode(inband)
exten => 7,1,AGI(turnoncoffeemachine.agi)
exten => 9,1,System(smbnuke boss)
Regards,

AA

_
Watch movie trailers online with the Xtra Broadband Channel  
http://xtra.co.nz/broadband

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