Re: [asterisk-users] Stable-Stable Asterisk

[Tzafrir Cohen]

On Thu, Aug 23, 2007 at 07:16:57PM -0400, Lee Jenkins wrote:
> Steve Totaro wrote:
> > David Gomillion wrote:
> >> On 8/23/07, *Ed Pastore* <[EMAIL PROTECTED] 
> >> > wrote:
> >>
> >> Hi, folks.
> >>
> >> I've been on the Asterisk Announce list for a while now, and it seems
> >> to me that the release versions of Asterisk are a bit bleeding-edge.
> >> They qualify as stable, but I wouldn't call them "production stable"
> >> since half the time a new one comes out, a fix for it comes out the
> >> next day.
> >>
> >>
> >> That's the niche that ABE is supposed to fill. I personally don't use 
> >> it, though. I just test the features I plan to use, disable everything 
> >> else, and seem to do OK.

What version of Asterisk is "current ABE" (something that would get
installed on a new system with no relation to other systems) based on?

> >>
> >>
> > 
> > I stay with 1.2.12 or somewhere around there.  "End Of Life" but seems 
> > to have a better ticker than 1.4.
> > 
> > Thanks,
> > Steve
> > 
> 
> 1.2.12/14/17 all have seemed very stable to me so far.

Both of which are anecdotial evidences.

Now suppose I had a major stability issue with 1.2.14 which was solved
with 1.2.18 (or 1.4.1). I would simply be dropped off that tatistics.
You'l be just left with those for which "1.2 works better".

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Prompt

[Tzafrir Cohen]

On Thu, Aug 23, 2007 at 03:13:40PM -0700, bilal ghayyad wrote:
> Dear Mojo;
> 
> Thanks for your help.
> 
> Why you said export ASTERISK_PROMPT="new prompt >"?

To make that a new value for the environment variable. 
An alternative method is:

ASTERISK_PROMPT="new prompt >" asterisk -r

There are actually some special % values there:

ASTERISK_PROMPT='date(d): %d, h(hostname) : %h, H(short hostname): %H, l(load 
avg): %l, s(system name) %s, t(time): %t, literal: %% > ' rasterisk

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Speech Rec on Voicemail

[mitcheloc]

Nuance offers an SDK to do something similar, I think they say you can
only expect between 45-60% accuracy using it though. Total cost is
about $6K to $8K for one server license.

If there are enough people interested in pooling money I'd be willing
to help set up a system to process voicemails and provide the Nuance
converted transcript. However, I figure the low accuracy would be the
biggest turn off from using Nuance.


On 8/23/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
> Ryan M. Colbert wrote:
> > I've had requests to processes incoming voicemails with voice
> > recognition routine and add the output text to the body of the email
> > message from * with the attached .wav file.  Has anyone implemented this
> > type of feature and willing to share some notes?
>
> I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this
> not too long ago.
>
> I get requests like this all the time -- but the technology is very far
> from being there.
>
> -Stephen-
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 

Mitchel Constantin
Snap - A desktop user interface for Asterisk
www.snapanumber.com

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] xPL and Asterisk?

[Jay Milk]

Matthew Rubenstein wrote:
>   I tried asking in another thread this week, but I'm not sure people saw
> the actual subject of the question. Does anyone know where to find
> documentation of xPL, the home automation interface? Specifically for
> integrating it with Asterisk. xPL is part of Trixbox, so it's being
> used, but where is some expertise for using it without Trixbox?
>   
http://www.google.com/search?q=xpl+home+automation

1st and 3rd results.

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk

[Zane C.B.]

On Tue, 21 Aug 2007 09:50:57 -0400
Dave Fullerton <[EMAIL PROTECTED]> wrote:

> Zane C.B. wrote:
> > On Tue, 21 Aug 2007 07:33:23 +0530
> > "Vidura Senadeera" <[EMAIL PROTECTED]> wrote:
> > 
> >> Dear All,
> >>
> >> I would like to get community's feedback with regard to RAID1
> >> ( Software or Hardware) implementations with asterisk.
> >>
> >> This is my setup
> >>
> >> Motherboard with SATA RAID1 support
> >> CENT OS 4.4
> >> Asterisk 1.2.19
> >> Libpri/zaptel latest release
> >> 2.8 Ghz Intel processor
> >> 2 80 GB SATA Hard disks
> >> 256 MB RAM
> >> digium PRI/E1 card
> >>
> >> Following are the concerns I am having
> >>
> >> I'm planing to put this asterisk server in production enviorment
> >> which is having E1 connection to the asterisk server,
> >> approximately 20 con-current calls, Music on hold, voice mail
> >> boxes.
> >>
> >> 1. If I use Software RAID, what would be the impact to my
> >> deployment? ( problems that I have to face with regard to the
> >> call flow ) 2. If I use Hardware based RAID 1, what would be the
> >> impact to the system? 3. According to your practical experiance
> >> what is the ideal solution among both options?
> >>
> >> I will be highly appreciate your feedback on this regard.
> > 
> > 1: Software RAID on Linux is way less than impressive. Plus last
> > a I checked Linux can't handle mirroring a entire disk. Last I
> > looked at it around a year ago you were limited to only mirroring
> > partitions, which is a joke from a administrative standpoint.
> > 2: No real impact other than a bad disk won't mean a reinstall.
> > 3: On Linux, go hardware. On FreeBSD it is personal choice.
> 
> You can (sort of) run raid on an entire disk, but you have to use
> LVM. You basically create a single partition on the disk, run raid
> on that partition and then use LVM with the /dev/md? device as a
> physical volume that you can then "partition" with LVM.

Yeah, still a ugly solution though.

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk

[Zane C.B.]

On Wed, 22 Aug 2007 12:37:26 -0600
Stephen Bosch <[EMAIL PROTECTED]> wrote:

> Zane C.B. wrote:
> > 1: Software RAID on Linux is way less than impressive. Plus last
> > a I checked Linux can't handle mirroring a entire disk. Last I
> > looked at it around a year ago you were limited to only mirroring
> > partitions, which is a joke from a administrative standpoint.
> 
> How is this any different in FreeBSD?
> 
> Could you explain to me how else you are going to mirror an entire
> disk in software when your boot partition is on the disk?

The raid info is done the same as on other decent system, it is stored
at the in the last sector of the provider.

making a mirrored freebsd system is like this...
1: install freebsd
2: dd if= of=<2nd drive for mirror>
3: gmirror label  <2nd drive>
4: mount 2nd drive and edit fstab to boot
using /dev/gmirror/
5: boot from 2nd drive
6: gmirror insert  


/me loves GEOM, the goddess of all disk subsystems or whatever.

http://www.freebsd.org/cgi/man.cgi?query=gmirror&apropos=0&sektion=0&manpath=FreeBSD+6.2-RELEASE&format=html

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it posible for an incoming to ring to Polycom and cell at the same time?

[Andrew Kohlsmith]

On Thursday 23 August 2007 11:22:23 pm Stephen Bosch wrote:
> > dial(SIP/polycom-on-my-desk&Local/5551212,15,tr)
> Will this work even if the Local is pointing to a Zap channel?
> As far as I know, this only works with SIP or IAX outgoing.

I'm not sure where you are getting that assumption from, as I have been 
Dialing Zap/foo&Zap/bar, SIP/foo&SIP/bar, IAX/foo&IAX/bar and combinations of 
all three for the past several years.

The only trick, as Anthony already showed, is to use 'r' when dialing a cell 
phone so that the caller hears the expected ringback, and not the 
carrier's "The cell phone you are trying to reach is out of the area" 
messages if the cell is out of range or off.

-A.

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it posible for an incoming to ring to Polycom and cell at the same time?

[Stephen Bosch]

Anthony Francis wrote:
> dial(SIP/polycom-on-my-desk&Local/5551212,15,tr)

Will this work even if the Local is pointing to a Zap channel?

As far as I know, this only works with SIP or IAX outgoing.

-Stephen-


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Speech Rec on Voicemail

[Stephen Bosch]

Ryan M. Colbert wrote:
> I’ve had requests to processes incoming voicemails with voice
> recognition routine and add the output text to the body of the email
> message from * with the attached .wav file.  Has anyone implemented this
> type of feature and willing to share some notes?

I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this
not too long ago.

I get requests like this all the time -- but the technology is very far
from being there.

-Stephen-

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk as a softswitch

[Russell Bryant]

Clayton Milos wrote:
> It is really a soft switch. Not a good one for high carrier class telco 
> usage I'd say but just fine for office PBX replacement, which is what it was 
> designed for. What it is missing AFAIK that a carrier class switch has to 
> have is a SS7 stack. The world's TDM exchanges use SS7 to route calls and 
> most carrier class soft switches can do SS7 over ip.

chan_zap in Asterisk trunk has SS7 support using libss7, a library written by
Matthew Fredrickson, who works at Digium.  Some people are already using it by
running trunk.  It will be in Asterisk 1.6.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] unable to load chan_unicall.so

[Moises Silva]

Edit logger.conf and learn how to enable debugging,verbose and all
kind of messages. Enable all levels of messages, try again and tell us
what is the error message exactly.

Regards,

On 8/23/07, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> Hi,
>  I am using debian 4.0 with version 2.6.18-4-686
>   I have downloaded the required files form site
> asterisk-1.2.24.tar.gz
> libmfcr2-0.0.3-1.4.tar.bz2
> libsupertone-0.0.2.tar.gz
> libunicall-0.0.3-1.4.tar.bz2
> spandsp-20060903.tar.gz
>
> I downloaded and installed the files in the follwing sequence
> spandsp
> libsupertone
> libunicall
> libmfcr2-0.0.3 is giving a lot of definition error
> I converted .src.rpm file of libmfcr2  to .deb file and installed it.
>
>the copying the chn_unicall.c and channels_Makefile.patch to
> channels subdirectory of asterisk-1.2.24
> but when I run ,asterisk -vvgc' on command line it gives error unable to
> load chan_unicall.so, but it is present in
> /usr/lib/asterisk/modules.
> Can anybody tell me how to trobleshoot it.
>
>
>  Can anybody tell me what to do how to remove this.
> Thanka and regards
> sanchal
>
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
"Within C++, there is a much smaller and cleaner language struggling
to get out."

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk as a softswitch

[James Jones]

Yes you could, but asterisk was designed to be a PBX. I would not use it as
soft switch due its limitations. It really depends on how much traffic you
are going to be passing.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Friday, 24 August 2007 1:11 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk as a softswitch

Mark Quitoriano wrote:
> Can i use asterisk as a softswitch?
This thread has been discussed over and over.  Search the archives, 
there are more thoughts and opinions there than you probably have time 
or desire to read.

Thanks,
Steve Totaro

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007
4:04 p.m.
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007
4:04 p.m.
 


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk as a softswitch

[Steve Totaro]

Mark Quitoriano wrote:
> Can i use asterisk as a softswitch?
This thread has been discussed over and over.  Search the archives, 
there are more thoughts and opinions there than you probably have time 
or desire to read.

Thanks,
Steve Totaro

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk as a softswitch

[Clayton Milos]

> Probably.
>
> PaulH
>
>
> On Fri, 2007-08-24 at 06:55 +0800, Mark Quitoriano wrote:
>> Can i use asterisk as a softswitch?

It is really a soft switch. Not a good one for high carrier class telco 
usage I'd say but just fine for office PBX replacement, which is what it was 
designed for. What it is missing AFAIK that a carrier class switch has to 
have is a SS7 stack. The world's TDM exchanges use SS7 to route calls and 
most carrier class soft switches can do SS7 over ip.

Here's a link that defines a soft switch according to the International 
Softswitch Consortium:
http://www.pcmag.com/encyclopedia_term/0,2542,t=softswitch&i=51659,00.asp

-Clay 


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk as a softswitch

[Paul Hales]

Probably.

PaulH


On Fri, 2007-08-24 at 06:55 +0800, Mark Quitoriano wrote:
> Can i use asterisk as a softswitch?
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] xPL and Asterisk?

[Matthew Rubenstein]

I tried asking in another thread this week, but I'm not sure people saw
the actual subject of the question. Does anyone know where to find
documentation of xPL, the home automation interface? Specifically for
integrating it with Asterisk. xPL is part of Trixbox, so it's being
used, but where is some expertise for using it without Trixbox?
-- 

(C) Matthew Rubenstein


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stable-Stable Asterisk

[Lee Jenkins]

Steve Totaro wrote:
> David Gomillion wrote:
>> On 8/23/07, *Ed Pastore* <[EMAIL PROTECTED] 
>> > wrote:
>>
>> Hi, folks.
>>
>> I've been on the Asterisk Announce list for a while now, and it seems
>> to me that the release versions of Asterisk are a bit bleeding-edge.
>> They qualify as stable, but I wouldn't call them "production stable"
>> since half the time a new one comes out, a fix for it comes out the
>> next day.
>>
>>
>> That's the niche that ABE is supposed to fill. I personally don't use 
>> it, though. I just test the features I plan to use, disable everything 
>> else, and seem to do OK.
>>
>>
> 
> I stay with 1.2.12 or somewhere around there.  "End Of Life" but seems 
> to have a better ticker than 1.4.
> 
> Thanks,
> Steve
> 

1.2.12/14/17 all have seemed very stable to me so far.

-- 
Warm Regards,

Lee

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Linksys (PAP2) delay time between hung up and line release

[Ramiro Gonzalez]

I have a PAP2 with 2 phone ports.
When I call them everything works fine until I hung up the call. There
is about 30-40 seconds until I can call to that extension again.
Before that it gives me busy messages.

Extension config:

exten => 199,1,Dial(SIP/199,30)
exten => 199,102,Hangup

Any suggestions?
Thanks

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] asterisk as a softswitch

[Mark Quitoriano]

Can i use asterisk as a softswitch?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] meetme conference problem

[Mark Quitoriano]

On 8/24/07, ram <[EMAIL PROTECTED]> wrote:
>
>
>
> On 8/23/07, Mark Quitoriano <[EMAIL PROTECTED]> wrote:
> >
> > Hi,
> >
> > im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
> > meetme conference,
> >
> > when i try to call meetme i get this from the asterisk console
> >
> > Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No
> > application 'MeetMe' for extension (sample, 65000, 1)
> >
> >
> > i recompiled my zaptel and asterisk, but the app_meetme file still
> > didn't install, what am i missing here?
>
>
>
> check meetme.conf
>



i don't know what's the problem, when i installed 1.2.20.1 zaptel everything
works.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stable-Stable Asterisk

[Steve Totaro]

David Gomillion wrote:
> On 8/23/07, *Ed Pastore* <[EMAIL PROTECTED] 
> > wrote:
>
> Hi, folks.
>
> I've been on the Asterisk Announce list for a while now, and it seems
> to me that the release versions of Asterisk are a bit bleeding-edge.
> They qualify as stable, but I wouldn't call them "production stable"
> since half the time a new one comes out, a fix for it comes out the
> next day.
>
>
> That's the niche that ABE is supposed to fill. I personally don't use 
> it, though. I just test the features I plan to use, disable everything 
> else, and seem to do OK.
>
>

I stay with 1.2.12 or somewhere around there.  "End Of Life" but seems 
to have a better ticker than 1.4.

Thanks,
Steve


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Prompt

[bilal ghayyad]

Dear Mojo;

Thanks for your help.

Why you said export ASTERISK_PROMPT="new prompt >"?

Regards
Bilal


I'm not sure what features/variables you can use, or
where to find 
information about that, but what this basically means
is you can change
 
your CLI prompt by this:

export ASTERISK_PROMPT="new prompt >"

then, what you access the CLI, instead of:

hostname*CLI>
you get
new prompt >

Moj

bilal ghayyad wrote:
> Hi List;
> 
> I read the following sentence:
> 
> "The CLI prompt is set with the ASTERISK_PROMPT UNIX
> environment variable"
> 
> In the following link:
> 
> http://www.voip-info.org/wiki/index.php
> page=Asterisk+CLI+prompt
> 
> The question is: what is the ASTERISK_PROMPT UNIX
> environment variable and where I can access it to
> change it? Also where I can find information about
it?
> 
> Regards
> Bilal Ghayad


  

Park yourself in front of a world of choices in alternative vehicles. Visit the 
Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/ 

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it posible for an incoming to ring to Polycom and cell at the same time?

[Anthony Francis]

dial(SIP/polycom-on-my-desk&Local/5551212,15,tr)

as an example.

Bob Gibson wrote:
>
> If it is posible for a imcoming call to ring both the Polycom desk
> phone and my cell phone at the same time, if I dont answer fall
> back to my voice mail box.
>
> I would like to hire someone to cofigure that for me.
>
> Bob
>
>  
>
>  
>
>
> -- 
> We've Got Your Name at Mail.com 
> 
> Get a *FREE* E-mail Account Today - Choose From 100+ Domains
> 
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TDM400P FXO click sounds

[ggonzalez]

1- I've tried running fxotune 
2- I've tried turning off all un-necessary hardware in the BIOS
3- I've tried on a different PCI slot. 
4- I've tried these suggestions:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting 
5- How I check if it the clicking and popping correlates to hard drive activity
?
6- I've not tried installing this board in another PC to test my FXOs 
7- I've an MSI motherboard and AMD athlon 64 x2 Dual core processor
8- I've Turning off echotraining.

Thanks for any suggest to solve this issue.






___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Is it posible for an incoming to ring to Polycom and cell at the same time?

[Bob Gibson]

  If it is posible for a imcoming call to ring both the Polycom desk
  phone and my cell phone at the same time, if I dont answer fall back
  to my voice mail box.

  I would like to hire someone to cofigure that for me.

  Bob

-- 
We've Got Your Name at http://www.mail.com !
Get a FREE E-mail Account Today - Choose From 100+ Domains

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Speech Rec on Voicemail

[David Gomillion]

On 8/23/07, Ryan M. Colbert <[EMAIL PROTECTED]> wrote:
>
>  I've had requests to processes incoming voicemails with voice recognition
> routine and add the output text to the body of the email message from * with
> the attached .wav file.  Has anyone implemented this type of feature and
> willing to share some notes?
>

That would be very interesting to see, if you get it working. Last I
checked, though, speech-to-text didn't work very well without a very small
language to choose from, far smaller than English.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stable-Stable Asterisk

[David Gomillion]

On 8/23/07, Ed Pastore <[EMAIL PROTECTED]> wrote:
>
> Hi, folks.
>
> I've been on the Asterisk Announce list for a while now, and it seems
> to me that the release versions of Asterisk are a bit bleeding-edge.
> They qualify as stable, but I wouldn't call them "production stable"
> since half the time a new one comes out, a fix for it comes out the
> next day.


That's the niche that ABE is supposed to fill. I personally don't use it,
though. I just test the features I plan to use, disable everything else, and
seem to do OK.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stable-Stable Asterisk

[Doug Lytle]

Ed Pastore wrote:
> since half the time a new one comes out, a fix for it comes out the  
> next day.
>
> So... that said, what's a good version to linger on? I don't *need*  
>
>   
Until 1.4 improves, I'm staying with 1.2

> I do know that I'm running some version of 1.2, and am also not sure  
> if I should stay there, or move up to 1.4.
>   

Show version

Doug

-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stable-Stable Asterisk

[Anthony Francis]

Show version from the CLI.

Ed Pastore wrote:
> Hi, folks.
>
> I've been on the Asterisk Announce list for a while now, and it seems  
> to me that the release versions of Asterisk are a bit bleeding-edge.  
> They qualify as stable, but I wouldn't call them "production stable"  
> since half the time a new one comes out, a fix for it comes out the  
> next day.
>
> So... that said, what's a good version to linger on? I don't *need*  
> anything particularly fancy, feature-wise, but would like to keep it  
> as secure and stable as possible. And I certainly don't mind fancy  
> features. :)
>
> Also (please forgive a newbie), how can I tell what version of  
> Asterisk I'm running? My current install was set up by a vendor and  
> I'm still learning the ropes. Where's the best place to look to find  
> the build number?
>
> I do know that I'm running some version of 1.2, and am also not sure  
> if I should stay there, or move up to 1.4.
>
> Thanks!
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Speech Rec on Voicemail

[Ryan M. Colbert]

I've had requests to processes incoming voicemails with voice recognition 
routine and add the output text to the body of the email message from * with 
the attached .wav file.  Has anyone implemented this type of feature and 
willing to share some notes?

Thanks!


Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue & McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
(407) 517-3105 - Direct Telephone
(407) 839-0120 - Main Office
(407) 841-9726 - Fax
http://www.rissman.com/

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Stable-Stable Asterisk

[Ed Pastore]

Hi, folks.

I've been on the Asterisk Announce list for a while now, and it seems  
to me that the release versions of Asterisk are a bit bleeding-edge.  
They qualify as stable, but I wouldn't call them "production stable"  
since half the time a new one comes out, a fix for it comes out the  
next day.

So... that said, what's a good version to linger on? I don't *need*  
anything particularly fancy, feature-wise, but would like to keep it  
as secure and stable as possible. And I certainly don't mind fancy  
features. :)

Also (please forgive a newbie), how can I tell what version of  
Asterisk I'm running? My current install was set up by a vendor and  
I'm still learning the ropes. Where's the best place to look to find  
the build number?

I do know that I'm running some version of 1.2, and am also not sure  
if I should stay there, or move up to 1.4.

Thanks!

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Someone please explain FXO/FXS module/channel relationships in Zaptel configuration to me

[Mojo with Horan & Company, LLC]

 From memory, your zaptel and zapata files look ok.  signalling for an 
FXO module would be FXS, and vice versa.  As far as I can tell, you're 
ok there.

Now, it's the FXO card that plugs into the phone line.  The FXS card 
gets a phone hooked up to it.  Dialing the phone would be
   Dial(Zap/1...
and dialing out the phone LINE would be
   Dial(Zap/2/18005551212...
for example, to dial 1 800 555 1212

Moj

Robert La Ferla wrote:
> Please explain the relationship between modules from the driver  
> (wctdm), the /etc/zaptel.conf file and zapata.conf.  Specifically, if  
> I have a FXS module 0 and FXO module 1, what should be used in  
> zaptel.conf and what should be used in zapata.conf?  Then finally, in  
> extensions.conf, what is the Zap channel for dialing out?  Zap/?
> 
> 
> % dmesg
> Module 0: Installed -- AUTO FXS/DPO
> Module 1: Installed -- AUTO FXO (FCC mode)
> Module 2: Not installed
> Module 3: Not installed
> Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules)
> 
> % cat /etc/zaptel.conf
> fxoks=1
> fxsks=2
> 
> % cat zapata.conf
> ...
> signalling=fxo_ks
> context=outgoing-analog
> echocancel=yes
> callerid=asreceived
> channel => 1
> 
> signalling=fxs_ks
> context=incoming-analog
> echocancel=yes
> callerid=asreceived
> channel => 2
> 
> 
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TC400B and show transcoder

[Kevin P. Fleming]

Ben Dinnerville wrote:

> The problem occurs when we have external (pstn) calls coming into / out 
> of the system (via an iax trunk), in which case we have no control over 
> frame size, as well as occurring with handsets directly connected to the 
> system.

Please contact Digium Support to work through these problems, as you
have unlimited installation support with the purchase of the product.

No, the TC400B does not provide any Zaptel 'spans', so it does not
provide a timing source.

What documentation is referring to the 'show transcoder' command? That
command is not in either Asterisk 1.2 or 1.4, so we need to get that
documentation fixed...

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Prompt

[Mojo with Horan & Company, LLC]

I'm not sure what features/variables you can use, or where to find 
information about that, but what this basically means is you can change 
your CLI prompt by this:

export ASTERISK_PROMPT="new prompt >"

then, what you access the CLI, instead of:

hostname*CLI>
you get
new prompt >

Moj

bilal ghayyad wrote:
> Hi List;
> 
> I read the following sentence:
> 
> "The CLI prompt is set with the ASTERISK_PROMPT UNIX
> environment variable"
> 
> In the following link:
> 
> http://www.voip-info.org/wiki/index.php
> page=Asterisk+CLI+prompt
> 
> The question is: what is the ASTERISK_PROMPT UNIX
> environment variable and where I can access it to
> change it? Also where I can find information about it?
> 
> Regards
> Bilal Ghayad
> 
> 
> 
>   
> 
> Park yourself in front of a world of choices in alternative vehicles. Visit 
> the Yahoo! Auto Green Center.
> http://autos.yahoo.com/green_center/ 
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] meetme conference problem

[ram]

On 8/23/07, Mark Quitoriano <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
> meetme conference,
>
> when i try to call meetme i get this from the asterisk console
>
> Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No
> application 'MeetMe' for extension (sample, 65000, 1)
>
>
> i recompiled my zaptel and asterisk, but the app_meetme file still didn't
> install, what am i missing here?



check meetme.conf

ram
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Tzafrir Cohen]

On Thu, Aug 23, 2007 at 12:43:15PM -0600, Anthony Francis wrote:
> The problem with any of these choices is that they do not address 
> logfile rotation.

Because this can be done with the standard system logrotate, or even by
asterisk (if you trust it to that). Decently-packaged Asterisk comes
with log rotation configuration.

Indeed the output of the CLI should not be simply logged. Asterisk has a
good enough logging facility that need not be replicated. There is no
need to start asterisk vebosely by default and spend useless CPU time on
useless messages. Use verbose messages when trying to debug a problem.
Let errors stand out when they come.

> 
> Try this:
> 
> http://cr.yp.to/daemontools.html

and endure the voodoo.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk Prompt

[bilal ghayyad]

Hi List;

I read the following sentence:

"The CLI prompt is set with the ASTERISK_PROMPT UNIX
environment variable"

In the following link:

http://www.voip-info.org/wiki/index.php
page=Asterisk+CLI+prompt

The question is: what is the ASTERISK_PROMPT UNIX
environment variable and where I can access it to
change it? Also where I can find information about it?

Regards
Bilal Ghayad



  

Park yourself in front of a world of choices in alternative vehicles. Visit the 
Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/ 

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] What is this?

[bilal ghayyad]

Hi List;

I saw this is written in that link:

http://www.voip-info.org/wiki/view/Asterisk+options

And really I was not able to understand for what is
that and where I can learn about it and how to write
such thing? Can some one advise me?


!/bin/bash 

asterisk Startup script for the asterisk PBX Server 

chkconfig: - 87 15 
description: Asterisk is a PBX server. 
processname: asterisk 
config: /etc/asterisk/ 
pidfile: /var/run/asterisk.pid 


Source function library. 
. /etc/rc.d/init.d/functions 

asterisk=/usr/sbin/asterisk 
prog=Asterisk 
pidfile=/var/run/asterisk.pid 
lockfile=/var/lock/subsys/asterisk 
RETVAL=0 

start() { 
   echo -n $"Starting $prog: " 
   daemon $asterisk $OPTIONS 
   RETVAL=$? 
   echo 
$RETVAL = 0  && touch ${lockfile} 
   return $RETVAL 
} 
stop() { 
   echo -n $"Stopping $prog: " 
   killproc $asterisk 
   RETVAL=$? 
   echo 
$RETVAL = 0  && rm -f ${lockfile} ${pidfile} 
} 
reload() { 
   echo -n $"Reloading $prog config files " 
   $asterisk -rx reload 
   RETVAL=$? 
   echo 
} 


See how we were called. 
case "$1" in 
 start) 
   start 
   ;; 
 stop) 
   stop 
   ;; 
 restart) 
   stop 
   start 
   ;; 
 reload) 
   reload 
   ;; 
 *) 
   echo $"Usage: $prog
{start|stop|restart|reload}" 
   exit 1 
esac 

exit $RETVAL 

Regards,
Bilal Ghayad


   

Need a vacation? Get great deals
to amazing places on Yahoo! Travel.
http://travel.yahoo.com/

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[James Collier]

You can configure logger.conf so that it will log just about everything you
could want.   

http://www.voip-info.org/wiki/index.php?page=Asterisk+config+logger.conf

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de
[EMAIL PROTECTED]
Enviado el: jueves, 23 de agosto de 2007 17:15
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] Asterisk Message Logs


Hello,

Is it possible to print the Asterisk message logs to a file, or is this
already done?  By message logs I mean the display that shows up on the
asterisk server when a call is made from one user to another.  I believe if
the verbosity is high, it can show what parts of the extension.conf file
that it uses when making the call.  I am trying to use two
Jain-sip-applet-phones, connected through an Asterisk server.  I can't seem
to get communication between the two phones.  Does anyone have any
experience using these open-source Jain-sip-applet-phones?

Thanks,

Denis


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
<>___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Anthony Francis]

The problem with any of these choices is that they do not address 
logfile rotation.

Try this:

http://cr.yp.to/daemontools.html

Ron Joffe wrote:
> On Thursday 23 August 2007 14:00, Tzafrir Cohen wrote:
>   
>>> I utilize this command:
>>>
>>> nohup script -f -c "asterisk -vvvTn" /tmp/asterisk.log &
>>>
>>> To start up my apps. This will log everything to a log file.
>>>   
>> Why nohup? And if you have nohup, why script?
>>
>> It will log everything until the cotrolling terminal is lost, right? I
>> think what you're actually looking for is screen.
>>
>>
>> If you want asterisk daemonized but still want it verbose, use -F
>> 
>
> I call asterisk startup from a shell script. "nohup" will guarantee that the 
> process will not die if the calling process (whatever started the shell 
> script) dies.
>
> script is what I use to make sure that everything that would otherwise go to 
> the asterisk cli output makes it into that file. We spawn our own extensions 
> from asterisk which the asterisk logging facility does not capture. This way 
> we get everything that would be seen on the cli. I'm not looking for screen 
> functionality.
>
> -F is not an option on my version of asterisk.
>
> Ron
>
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Ron Joffe]

On Thursday 23 August 2007 14:00, Tzafrir Cohen wrote:
> >
> > I utilize this command:
> >
> > nohup script -f -c "asterisk -vvvTn" /tmp/asterisk.log &
> >
> > To start up my apps. This will log everything to a log file.
>
> Why nohup? And if you have nohup, why script?
>
> It will log everything until the cotrolling terminal is lost, right? I
> think what you're actually looking for is screen.
>
>
> If you want asterisk daemonized but still want it verbose, use -F

I call asterisk startup from a shell script. "nohup" will guarantee that the 
process will not die if the calling process (whatever started the shell 
script) dies.

script is what I use to make sure that everything that would otherwise go to 
the asterisk cli output makes it into that file. We spawn our own extensions 
from asterisk which the asterisk logging facility does not capture. This way 
we get everything that would be seen on the cli. I'm not looking for screen 
functionality.

-F is not an option on my version of asterisk.

Ron



___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] channel not hungup (zombie?) so call limit not reset to zero

[Rizwan Hisham]

im having a strange problem related to call-limit for peers. well im not
sure if its related to call-limmit or not. Bottom line is:

I call a user A, from user B. user B hears silence, untill it goes to
voicemail. when user B hangsup. user B's call limit is reset to 0 but user
A's call limit is not reset.strange thing is user A's status on cli is shown
as NOANSWER, while user B did not even hear ringing. when i use "sip show
channels" command, it shows me a channel for user A like below:

crunch  30d926c1055  00102/0  unkn  No   Init: INVITE

It stays in INVITE state unless i restart my asterisk server. when i restart
the channel is clear (ofcorse)

So my guess is, its a zombiee channel which asterisk forgot to hangup. WHY?
i dont know, maybe there is a problem in sip signalling due to which
asterisk didnt recieve the bye signal in the first place or maybe its
asterisk fault totally.

So because it is not hungup by asterisk thats why its call limit is not
reset to zero. I dont have sip debug for this problem yet, i'll post it
later when i have it. meanwhile if somebody has experienced a similar
problem and has successfully fixed it, then plz share my burden and help me.
-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] meetme conference problem

[Tzafrir Cohen]

On Fri, Aug 24, 2007 at 12:16:24AM +0800, Mark Quitoriano wrote:
> Hi,
> 
> im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
> meetme conference,
> 
> when i try to call meetme i get this from the asterisk console
> 
> Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No
> application 'MeetMe' for extension (sample, 65000, 1)
> 
> 
> i recompiled my zaptel and asterisk, but the app_meetme file still didn't
> install, what am i missing here?

Do you have a zaptel timing source?

if head -c 0 /dev/zap/pseudo; then echo "I have a working timing source"; fi

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Kutman.DK]

Yes, any output from the console logs.  I tried viewing the full file and it 
looks like it's what I was looking for.  Thanks for the help.

Denis

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Jones
Sent: Thursday, August 23, 2007 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Message Logs


[EMAIL PROTECTED] wrote:
> Thanks for your reply.  I have previously looked at the logger.conf file.  I 
> see that the various types of information can be logged in different ways.  
> After setting the various information types with whatever I want logged, is 
> it possible to save the actual logs to a file (ie:  As the messages are bring 
> printed, save them all to a file to be viewed later).
>   
What do you mean by actual logs?  Console (CLI) output?

Brian.


> Thanks,
>
> Denis
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Jared Smith
> Sent: Thursday, August 23, 2007 12:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk Message Logs
>
>
> On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote:
>   
>> Is it possible to print the Asterisk message logs to a file, or is
>> this already done?  
>> 
>
> You want to look at the logger.conf configuration file, and see how your
> Asterisk system is set to log the various types of information (such as
> debug messages, verbose messages, DTMF messages, etc.) are logged.  
>
> After changing logger.conf, you can type "logger" reload at the Asterisk
> CLI to make the changes take effect.
>
>
>   


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Tzafrir Cohen]

On Thu, Aug 23, 2007 at 01:07:36PM -0400, Ron Joffe wrote:
> On Thursday 23 August 2007 12:46, [EMAIL PROTECTED] wrote:
> > Thanks for your reply.  I have previously looked at the logger.conf file. 
> > I see that the various types of information can be logged in different
> > ways.  After setting the various information types with whatever I want
> > logged, is it possible to save the actual logs to a file (ie:  As the
> > messages are bring printed, save them all to a file to be viewed later).
> 
> I utilize this command:
> 
> nohup script -f -c "asterisk -vvvTn" /tmp/asterisk.log &
> 
> To start up my apps. This will log everything to a log file.

Why nohup? And if you have nohup, why script?

It will log everything until the cotrolling terminal is lost, right? I
think what you're actually looking for is screen.


If you want asterisk daemonized but still want it verbose, use -F

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 1.4 Branch -- which revision

[Dave Fullerton]

Jay Milk wrote:
> I'm finally migrating from the 6/7/05 CVS version to 1.4.  Had quite a 
> run, I have to admit.  Asterisk itself only segfaulted once or twice, 
> but the dns issues have been bothering me.  And the box just needs to 
> go.  Everything is going on a Ubuntu 6.06TLS server, that's been 
> perfectly stable.  I had 1.4.1 installed and running, but not 
> configured.  Yesterday I upgraded to 1.4.11, which went smoothly... 
> alas, I really wanted chan_mobile.
> 
> I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that 
> didn't do it.  So it looks like I'd have to use the 1.4 branch for both 
> asterisk and addons.  What's the recommended revision here?  I don't 
> need bleeding edge (obviously), I just need it stable with chan_mobile 
> and not too much else.
> 
> Thanks!

If you just want chan_mobile, there was a message just yesterday that 
covered this:

Thomas Kenyon <[EMAIL PROTECTED]> wrote:
"Try checking out r421 of asterisk-addons, and replacing ast_debug(1,
with ast_log(LOG_DEBUG, in all instances in chan_mobile.c.

(Still only compile chan_mobile.c.

This appears to work with 421, but not 423."


I've done this myself with asterisk 1.4.10 and I was able to compile it 
and install it. I haven't been able to test it until I can borrow a 
phone with bluetooth.

-Dave

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 1.4 Branch -- which revision

[Jay Milk]

Jason Parker wrote:
> Jay Milk wrote:
>   
>> I'm finally migrating from the 6/7/05 CVS version to 1.4.  Had quite a 
>> run, I have to admit.  Asterisk itself only segfaulted once or twice, 
>> but the dns issues have been bothering me.  And the box just needs to 
>> go.  Everything is going on a Ubuntu 6.06TLS server, that's been 
>> perfectly stable.  I had 1.4.1 installed and running, but not 
>> configured.  Yesterday I upgraded to 1.4.11, which went smoothly... 
>> alas, I really wanted chan_mobile.
>>
>> I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that 
>> didn't do it.  So it looks like I'd have to use the 1.4 branch for both 
>> asterisk and addons.  What's the recommended revision here?  I don't 
>> need bleeding edge (obviously), I just need it stable with chan_mobile 
>> and not too much else.
>>
>> Thanks!
>>
>> 
>
> chan_mobile isn't in asterisk-addons in 1.4 - only trunk.  You'll likely have
> to backport it...  (it was developed against 1.4, so the diff from trunk is
> probably trivial)
>
>   
Hmm, I got myself confused into thinking I checked out the 1.4 branch 
somehow.  Or maybe that 22-1.4.4.patch file had partial success.  So, to 
restate the question --

Which trunk revision are folks using successfully?

(and no, a diff isn't trivial to someone who barely keeps a command 
prompt ahead of certain disaster ;-)


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Brian Jones]

[EMAIL PROTECTED] wrote:
> Thanks for your reply.  I have previously looked at the logger.conf file.  I 
> see that the various types of information can be logged in different ways.  
> After setting the various information types with whatever I want logged, is 
> it possible to save the actual logs to a file (ie:  As the messages are bring 
> printed, save them all to a file to be viewed later).
>   
What do you mean by actual logs?  Console (CLI) output?

Brian.


> Thanks,
>
> Denis
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Jared Smith
> Sent: Thursday, August 23, 2007 12:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk Message Logs
>
>
> On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote:
>   
>> Is it possible to print the Asterisk message logs to a file, or is
>> this already done?  
>> 
>
> You want to look at the logger.conf configuration file, and see how your
> Asterisk system is set to log the various types of information (such as
> debug messages, verbose messages, DTMF messages, etc.) are logged.  
>
> After changing logger.conf, you can type "logger" reload at the Asterisk
> CLI to make the changes take effect.
>
>
>   


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Brian Jones]

[EMAIL PROTECTED] wrote:
> Hello,
>
> Is it possible to print the Asterisk message logs to a file, or is this 
> already done?  By message logs I mean the display that shows up on the 
> asterisk server when a call is made from one user to another.  I believe if 
> the verbosity is high, it can show what parts of the extension.conf file that 
> it uses when making the call.  I am trying to use two Jain-sip-applet-phones, 
> connected through an Asterisk server.  I can't seem to get communication 
> between the two phones.  Does anyone have any experience using these 
> open-source Jain-sip-applet-phones?
>
> Thanks,
>
> Denis
>   
Add this to logger.conf:

full => notice,warning,error,debug,verbose

and you should have most of the output stored in /var/log/asterisk/full

Brian.


>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Ron Joffe]

On Thursday 23 August 2007 12:46, [EMAIL PROTECTED] wrote:
> Thanks for your reply.  I have previously looked at the logger.conf file. 
> I see that the various types of information can be logged in different
> ways.  After setting the various information types with whatever I want
> logged, is it possible to save the actual logs to a file (ie:  As the
> messages are bring printed, save them all to a file to be viewed later).

I utilize this command:

nohup script -f -c "asterisk -vvvTn" /tmp/asterisk.log &

To start up my apps. This will log everything to a log file.

Ron




___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Kutman.DK]

Thanks for your reply.  I have previously looked at the logger.conf file.  I 
see that the various types of information can be logged in different ways.  
After setting the various information types with whatever I want logged, is it 
possible to save the actual logs to a file (ie:  As the messages are bring 
printed, save them all to a file to be viewed later).

Thanks,

Denis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jared Smith
Sent: Thursday, August 23, 2007 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Message Logs


On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote:
> Is it possible to print the Asterisk message logs to a file, or is
> this already done?  

You want to look at the logger.conf configuration file, and see how your
Asterisk system is set to log the various types of information (such as
debug messages, verbose messages, DTMF messages, etc.) are logged.  

After changing logger.conf, you can type "logger" reload at the Asterisk
CLI to make the changes take effect.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 1.4 Branch -- which revision

[Jason Parker]

Jay Milk wrote:
> I'm finally migrating from the 6/7/05 CVS version to 1.4.  Had quite a 
> run, I have to admit.  Asterisk itself only segfaulted once or twice, 
> but the dns issues have been bothering me.  And the box just needs to 
> go.  Everything is going on a Ubuntu 6.06TLS server, that's been 
> perfectly stable.  I had 1.4.1 installed and running, but not 
> configured.  Yesterday I upgraded to 1.4.11, which went smoothly... 
> alas, I really wanted chan_mobile.
> 
> I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that 
> didn't do it.  So it looks like I'd have to use the 1.4 branch for both 
> asterisk and addons.  What's the recommended revision here?  I don't 
> need bleeding edge (obviously), I just need it stable with chan_mobile 
> and not too much else.
> 
> Thanks!
> 

chan_mobile isn't in asterisk-addons in 1.4 - only trunk.  You'll likely have
to backport it...  (it was developed against 1.4, so the diff from trunk is
probably trivial)

-- 
Jason Parker
Digium

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] 1.4 Branch -- which revision

[Jay Milk]

I'm finally migrating from the 6/7/05 CVS version to 1.4.  Had quite a 
run, I have to admit.  Asterisk itself only segfaulted once or twice, 
but the dns issues have been bothering me.  And the box just needs to 
go.  Everything is going on a Ubuntu 6.06TLS server, that's been 
perfectly stable.  I had 1.4.1 installed and running, but not 
configured.  Yesterday I upgraded to 1.4.11, which went smoothly... 
alas, I really wanted chan_mobile.

I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that 
didn't do it.  So it looks like I'd have to use the 1.4 branch for both 
asterisk and addons.  What's the recommended revision here?  I don't 
need bleeding edge (obviously), I just need it stable with chan_mobile 
and not too much else.

Thanks!

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to configure and use GCE4019VOIP phone using asterisk

[sanchal . singh]

Hi,
I have GCE4019VOIP IP phone with me. Can anybody tell me the steps
how to use it for communication in the LAN with other sip phones. I want
help
from the IP phone side as I have already done it with SIP soft
phone...
Thanks and Regards,
Sanchal


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] meetme conference problem

[Mark Quitoriano]

Hi,

im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
meetme conference,

when i try to call meetme i get this from the asterisk console

Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No
application 'MeetMe' for extension (sample, 65000, 1)


i recompiled my zaptel and asterisk, but the app_meetme file still didn't
install, what am i missing here?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Message Logs

[Jared Smith]

On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote:
> Is it possible to print the Asterisk message logs to a file, or is
> this already done?  

You want to look at the logger.conf configuration file, and see how your
Asterisk system is set to log the various types of information (such as
debug messages, verbose messages, DTMF messages, etc.) are logged.  

After changing logger.conf, you can type "logger" reload at the Asterisk
CLI to make the changes take effect.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [PHP-AGI] Problem executing script

[Karim H]

Problem resolved :Two things :1 - php agi instead of php-cli (I have apache 
running on my server too and it edits himself all the php.ini it finds)2 - 
Error using php-q it is amistake : php -q (of course =) )Their is a real 
problem with asterisk concerning errors in agi script.If there is an error in 
the script itself asterisk give back : No such file or directory even if the 
error is just that ";" is missing...Thanks for the helpFrom: [EMAIL PROTECTED]: 
[EMAIL PROTECTED]: Thu, 23 Aug 2007 14:51:27 +Subject: [asterisk-users] 
[PHP-AGI] Problem executing script





Hello,I have succeded in compiling and configuring My TDM Card and asterisk, 
all works fine. But I have a problem using the PHP Agi.The CLI tells me that 
when I call my number :-- Starting simple switch on 'Zap/4-1'-- Executing 
[EMAIL PROTECTED]:1] Answer("Zap/4-1", "") in new stack-- Executing [EMAIL 
PROTECTED]:2] AGI("Zap/4-1", "rabot.agi") in new stack-- Launched AGI 
Script /var/lib/asterisk/agi-bin/rabot.agi  ==  rabot.agi: Failed to execute 
'/var/lib/asterisk/agi-bin/rabot.agi': No such file or directory-- AGI 
Script rabot.agi completed, returning 0  == Auto fallthrough, channel 'Zap/4-1' 
status is 'UNKNOWN'-- Hungup 'Zap/4-1'I tought first that it was a problem 
of chmod so I change the chmod of all the directory agi-bin TO 777But it 
changed nothing. I have verify that php was well indicate at the beginning of 
the script :#!/usr/bin/php-qAnd there is a php exec at /usr/binAny ideas about 
this problem ?Thank for you helpKheraudBesoin d'un e-mail ? Créez gratuitement 
un compte Windows Live Hotmail et bénéficiez de 2 Go de stockage ! Windows Live 
Hotmail

_
Windows Live Messenger vous offre 30 nouvelles émoticônes gratuites, installées 
directement dans votre Messenger !
http://www.emoticones-messenger.fr/___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] libmfcr2 is giving definition error while compiling

[Patrick]

On Thu, 2007-08-23 at 20:16 +0530, [EMAIL PROTECTED]
wrote:
> Hi,
>  I am using debian 4.0 with version 2.6.18-4-686
>   I have downloaded the required files form site
>   asterisk-1.2.24.tar.gz
>   libmfcr2-0.0.3-1.4.tar.bz2
>   libsupertone-0.0.2.tar.gz
>   libunicall-0.0.3-1.4.tar.bz2
>   spandsp-20060903.tar.gz

I use spandsp-0.0.3. Try that and see if it solves your problem.


>   I downloaded and installed the files in the follwing sequence
>   spandsp
>   libsupertone
>   libunicall
>   Till here it is compiling and copying .so library to 
> /usr/local/lib/
> libmfcr2-0.0.3 is giving a lot of definition error
>   I converted .src.rpm file of libmfcr2  to .deb file and installed so .so
> files are not their in
>   /usr/local/lib

Not sure if I understand you correctly but can't you just manually copy
the .so files to the right directory where libmfcr2 can find them. Or
maybe you just need to run ldconfig so the libs in /usr/local/lib are
picked up correctly (check with ldconfig -v).

>   With some minor changes the libmfcr2 get compiled successfully but
> some rpath error
> was  coming.

I don't know what is causing the rpath problems. I don't get rpath
errors when the rpath check is done during the rpmbuild on a Fedora 7
box.

Regards,
Patrick


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zaptel 1.2.20.1 and 1.4.5.1 released

[Matthew Fredrickson]

Steve Kennedy wrote:
> On Wed, Aug 22, 2007 at 01:19:14PM -0500, Asterisk Development Team wrote:
> 
>> The Asterisk.org development team has announced the release of Zaptel 
>> versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in 
>> the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel 
>> releases, as well as a handful of other issues.  See the respective 
>> Changelogs for more details.
>> Both releases are available as a tarball as well as a patch against the 
>> previous release. They are available for download from downloads.digium.com.
> 
> Don't seem to be on www.asterisk.org (1.2.19 and 1.4.4)

Sorry, I still  have to get the powers that be to update the home page :-)

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [PHP-AGI] Problem executing script

[Mik Cheez]

Did you confirm that the file exists?

/var/lib/asterisk/agi-bin/rabot.agi

Also, in your script (wherever it actually is), put a space between php 
and -q

#!/usr/bin/php -q

Karim H wrote:
> Hello,
> I have succeded in compiling and configuring My TDM Card and asterisk, 
> all works fine.
> But I have a problem using the PHP Agi.
> The CLI tells me that when I call my number :
> 
> -- Starting simple switch on 'Zap/4-1'
> -- Executing [EMAIL PROTECTED]:1] Answer("Zap/4-1", "") in new stack
> -- Executing [EMAIL PROTECTED]:2] AGI("Zap/4-1", "rabot.agi") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/rabot.agi
>   ==  rabot.agi: Failed to execute 
> '/var/lib/asterisk/agi-bin/rabot.agi': No such file or directory
> -- AGI Script rabot.agi completed, returning 0
>   == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
> -- Hungup 'Zap/4-1'
> 
> I tought first that it was a problem of chmod so I change the chmod of 
> all the directory agi-bin TO 777
> 
> But it changed nothing. I have verify that php was well indicate at the 
> beginning of the script :
> #!/usr/bin/php-q
> 
> And there is a php exec at /usr/bin
> 
> Any ideas about this problem ?
> 
> Thank for you help
> 
> Kheraud
> 
> 
> Besoin d'un e-mail ? Créez gratuitement un compte Windows Live Hotmail 
> et bénéficiez de 2 Go de stockage ! Windows Live Hotmail 
> 
> 
> 
> 
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [PHP-AGI] Problem executing script

[Louis-Eric]

Hey there,

The file must be accessible by the process 
calling upon it. Try a chown using the Asterisk process user name.


Cheers,

Louis-Eric



At 09:51 AM 8/23/2007, Karim H wrote:

Hello,
I have succeded in compiling and configuring My 
TDM Card and asterisk, all works fine.

But I have a problem using the PHP Agi.
The CLI tells me that when I call my number :

-- Starting simple switch on 'Zap/4-1'
-- Executing [EMAIL PROTECTED]:1] Answer("Zap/4-1", "") in new stack
-- Executing [EMAIL PROTECTED]:2] AGI("Zap/4-1", "rabot.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/rabot.agi
  ==  rabot.agi: Failed to execute 
'/var/lib/asterisk/agi-bin/rabot.agi': No such file or directory

-- AGI Script rabot.agi completed, returning 0
  == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
-- Hungup 'Zap/4-1'

I tought first that it was a problem of chmod so 
I change the chmod of all the directory agi-bin TO 777


But it changed nothing. I have verify that php 
was well indicate at the beginning of the script :

#!/usr/bin/php-q

And there is a php exec at /usr/bin

Any ideas about this problem ?

Thank for you help

Kheraud


--
Besoin d'un e-mail ? Créez gratuitement un 
compte Windows Live Hotmail et bénéficiez de 2 
Go de stockage ! 
Windows Live Hotmail

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk Message Logs

[Kutman.DK]

Hello,

Is it possible to print the Asterisk message logs to a file, or is this already 
done?  By message logs I mean the display that shows up on the asterisk server 
when a call is made from one user to another.  I believe if the verbosity is 
high, it can show what parts of the extension.conf file that it uses when 
making the call.  I am trying to use two Jain-sip-applet-phones, connected 
through an Asterisk server.  I can't seem to get communication between the two 
phones.  Does anyone have any experience using these open-source 
Jain-sip-applet-phones?

Thanks,

Denis


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] [PHP-AGI] Problem executing script

[Karim H]

Hello,I have succeded in compiling and configuring My TDM Card and asterisk, 
all works fine. But I have a problem using the PHP Agi.The CLI tells me that 
when I call my number :-- Starting simple switch on 'Zap/4-1'-- Executing 
[EMAIL PROTECTED]:1] Answer("Zap/4-1", "") in new stack-- Executing [EMAIL 
PROTECTED]:2] AGI("Zap/4-1", "rabot.agi") in new stack-- Launched AGI 
Script /var/lib/asterisk/agi-bin/rabot.agi  ==  rabot.agi: Failed to execute 
'/var/lib/asterisk/agi-bin/rabot.agi': No such file or directory-- AGI 
Script rabot.agi completed, returning 0  == Auto fallthrough, channel 'Zap/4-1' 
status is 'UNKNOWN'-- Hungup 'Zap/4-1'I tought first that it was a problem 
of chmod so I change the chmod of all the directory agi-bin TO 777But it 
changed nothing. I have verify that php was well indicate at the beginning of 
the script :#!/usr/bin/php-qAnd there is a php exec at /usr/binAny ideas about 
this problem ?Thank for you helpKheraud
_
Windows Live Messenger vous offre 30 nouvelles émoticônes gratuites, installées 
directement dans votre Messenger !
http://www.emoticones-messenger.fr/___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] libmfcr2 is giving definition error while compiling

[sanchal . singh]

Hi,
 I am using debian 4.0 with version 2.6.18-4-686
  I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
libunicall-0.0.3-1.4.tar.bz2
spandsp-20060903.tar.gz

I downloaded and installed the files in the follwing sequence
spandsp
libsupertone
libunicall
Till here it is compiling and copying .so library to 
/usr/local/lib/
libmfcr2-0.0.3 is giving a lot of definition error
I converted .src.rpm file of libmfcr2  to .deb file and installed so .so
files are not their in
/usr/local/lib

  With some minor changes the libmfcr2 get compiled successfully but
some rpath error
was  coming.


 Can anybody tell me what to do how to remove this.
Thanka and regards
sanchal


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] libmfcr2 is giving definition error while compiling

[sanchal . singh]

Hi,
 I am using debian 4.0 with version 2.6.18-4-686
  I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
libunicall-0.0.3-1.4.tar.bz2
spandsp-20060903.tar.gz

I downloaded and installed the files in the follwing sequence
spandsp
libsupertone
libunicall
Till here it is compiling and copying .so library to 
/usr/local/lib/
libmfcr2-0.0.3 is giving a lot of definition error
I converted .src.rpm file of libmfcr2  to .deb file and installed so .so
files are not their in
/usr/local/lib
 Can anybody tell me what to do how to remove this.
Thanka and regards
sanchal





___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] unable to load chan_unicall.so

[sanchal . singh]

Hi,
 I am using debian 4.0 with version 2.6.18-4-686
  I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
libunicall-0.0.3-1.4.tar.bz2
spandsp-20060903.tar.gz

I downloaded and installed the files in the follwing sequence
spandsp
libsupertone
libunicall
libmfcr2-0.0.3 is giving a lot of definition error
I converted .src.rpm file of libmfcr2  to .deb file and installed it.

   the copying the chn_unicall.c and channels_Makefile.patch to
channels subdirectory of asterisk-1.2.24
but when I run ,asterisk -vvgc' on command line it gives error unable to
load chan_unicall.so, but it is present in
/usr/lib/asterisk/modules.
Can anybody tell me how to trobleshoot it.


 Can anybody tell me what to do how to remove this.
Thanka and regards
sanchal



___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [Serusers] why combine ser with asterisk

[SIP]

Offers them? Yes. Offers them in a clean, friendly, usable package? Not 
so much yet.

SEMS has raw capability, but if you want it to do many of the things 
Asterisk can do, you need to know how to code that yourself, or you're 
going to be digging about the code for documentation on features (since 
the current docs are not the world's greatest).

Don't get me wrong, SEMS has its place, and is a constantly evolving 
work of art (we use SEMS for several things in our environment), but 
comparing SEMS to Asterisk is a bit like comparing a bunch of car parts 
to a Porsche.

N.


Fredrik Lundmark wrote:
> I'm still learning myself, but SEMS (iptel.org/sems) seems to offer 
> many of the media- and/or b2bua-functions that Asterisk do.
>
> ///Fredrik
>
>
>
> - Original Message - From: "SIP" <[EMAIL PROTECTED]>
> To: "Nhadie" <[EMAIL PROTECTED]>
> Cc: ; <[EMAIL PROTECTED]>
> Sent: Thursday, August 23, 2007 1:38 PM
> Subject: Re: [Serusers] why combine ser with asterisk
>
>
>> Asterisk is an excellent PBX system, and makes a very good endpoint in
>> the SIP chain for all sorts of things -- IVR systems, voicemail
>> applications, automated messages, etc.
>>
>> It has an extremely well-written CDR engine, so many people mesh it with
>> billing applications to produce accurate accounting information. It also
>> is fully aware of the media stream, which means it's capable of cutting
>> off a call mid-stream, injecting audio into the call, etc, etc.
>>
>> Programming for Asterisk addons can be easily done in just about any
>> language, and it meshes well with the overall structure. Programming for
>> SER is... not so simple.
>>
>> As for running them both on the same box, the biggest problem would be
>> resources. Unlike SER, Asterisk is not designed to be a carrier-grade
>> SIP proxy. If you're actually proxying the media stream, you'd be
>> hard-pressed to squeeze more than 150 simultaneous calls out of Asterisk
>> on even the beefiest of hardware. Add SER to the same box, and you will
>> quickly run into resource problems in medium-sized environments. It also
>> doesn't have a lot of the SIP proxy functionality that SER has.
>>
>> If you're careful, you can configure Asterisk NOT to handle the media
>> stream and still use it for prepaid solutions (using astcc or
>> asterisk-b2bua), and this will save you bandwidth (but you'll still
>> likely run into NAT issues that need to be dealt with somehow) and still
>> let you use Asterisk as an in-between point.
>>
>> Together, Asterisk and SER make a very powerful combination for
>> providing a full suite of services to clientele, and each plays well off
>> the other's strengths.
>>
>> N.
>>
>>
>>
>> Nhadie wrote:
>>> Hi All,
>>>
>>> What's the advantage of combining ser with asterisk? I always see
>>> comments like using ser with asterisk is a very good solution etc. etc.
>>> the thing i liked with ser is that it does not do codec translation,
>>> which saves me cpu usage and also bandwidth. if i combine it with
>>> asterisk, would it not use codec translation?
>>>
>>> i also read that there is also a problem when ser and asterisk is 
>>> run on
>>> the same machine, why is it so?
>>> if use prepaid billing solution for asterisk like astcc, would i 
>>> then be
>>> able to provide prepaid service?
>>>
>>> soryy for asking too much, i'd just like to really understand it. Thank
>>> You in advanced.
>>>
>>> Regards,
>>> Nhadie
>>> ___
>>> Serusers mailing list
>>> [EMAIL PROTECTED]
>>> http://lists.iptel.org/mailman/listinfo/serusers
>>>
>>
>> ___
>> Serusers mailing list
>> [EMAIL PROTECTED]
>> http://lists.iptel.org/mailman/listinfo/serusers
>>


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] contact header is missing in 200OK for SUBSCRIBE

[sumanth achar]

Hi,
Hi,
 I am trying to SUBSCRIBE for message waiting indications to asterisk,
it sends 200 OK but contact header is missing(it is mandatory since
subscribe is dialog establishing method), due to which parsing fails and
also expires is 0 in the 200 OK  any body knows about these issue...?



On 8/23/07, sumanth achar <[EMAIL PROTECTED]> wrote:
>
> Hi,
>  I am trying to SUBSCRIBE for message waiting indications to asterisk,
> it sends 200 OK but contact header is missing(it is mandatory since
> subscribe is dialog establishing method), due to which parsing fails, any
> body knows about this issue...?
>
> Regards,
> Subramanya
>
>
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [Serusers] why combine ser with asterisk

[SIP]

Asterisk is an excellent PBX system, and makes a very good endpoint in 
the SIP chain for all sorts of things -- IVR systems, voicemail 
applications, automated messages, etc.

It has an extremely well-written CDR engine, so many people mesh it with 
billing applications to produce accurate accounting information. It also 
is fully aware of the media stream, which means it's capable of cutting 
off a call mid-stream, injecting audio into the call, etc, etc. 

Programming for Asterisk addons can be easily done in just about any 
language, and it meshes well with the overall structure. Programming for 
SER is... not so simple.

As for running them both on the same box, the biggest problem would be 
resources. Unlike SER, Asterisk is not designed to be a carrier-grade 
SIP proxy. If you're actually proxying the media stream, you'd be 
hard-pressed to squeeze more than 150 simultaneous calls out of Asterisk 
on even the beefiest of hardware. Add SER to the same box, and you will 
quickly run into resource problems in medium-sized environments. It also 
doesn't have a lot of the SIP proxy functionality that SER has.

If you're careful, you can configure Asterisk NOT to handle the media 
stream and still use it for prepaid solutions (using astcc or 
asterisk-b2bua), and this will save you bandwidth (but you'll still 
likely run into NAT issues that need to be dealt with somehow) and still 
let you use Asterisk as an in-between point.

Together, Asterisk and SER make a very powerful combination for 
providing a full suite of services to clientele, and each plays well off 
the other's strengths.

N.



Nhadie wrote:
> Hi All,
>
> What's the advantage of combining ser with asterisk? I always see 
> comments like using ser with asterisk is a very good solution etc. etc.
> the thing i liked with ser is that it does not do codec translation, 
> which saves me cpu usage and also bandwidth. if i combine it with 
> asterisk, would it not use codec translation?
>
> i also read that there is also a problem when ser and asterisk is run on 
> the same machine, why is it so?
> if use prepaid billing solution for asterisk like astcc, would i then be 
> able to provide prepaid service?
>
> soryy for asking too much, i'd just like to really understand it. Thank 
> You in advanced.
>
> Regards,
> Nhadie
> ___
> Serusers mailing list
> [EMAIL PROTECTED]
> http://lists.iptel.org/mailman/listinfo/serusers
>   


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cisco firmwares 3.6.3 vs 3.8.6

[Adrian Marsh]

Thanks for that Arnaud,  saw it myself this morning, but the download
link takes me to a "page not found" cisco page :(  I've reported it on
their broken links page...


Adrian Marsh
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arnaud
Ligot
Sent: 22 August 2007 13:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco firmwares 3.6.3 vs 3.8.6

FYI about cisco firmware:
http://www.cisco.com/warp/public/707/cisco-sr-20070821-sip.shtml


A.


On Wed, 2007-08-22 at 12:26 +0100, Adrian Marsh wrote:
> Hi All,
> 
> A question for those with Cisco 7940/60 SIP phones.  I used to load
> POS3-06-03-00 Firmware to the cisco phones.  A month or so ago, I ran
> some tests and found that latest 3.8.6 firmware worked well, and
solved
> an issue or two on the phones.
> 
> I've a number of users who work outside of the LAN.  Our phones use
DNS
> names to talk to A*k, so in theory, just enabling NAT makes the phone
> work outside the LAN (home users, remote users, etc).  However, when
we
> loaded the 3.8.6 firmware to these phones, we've found the phones no
> longer work outside of the LAN.  Using Etherreal, we've found that the
> communication between the Phone and A*k breaks (A*k never sees the
> Register packets, but the phone does seem to send them.  I'll post
more
> detail if its needed, but I wondered if anyone else has seen this ?
The
> size of the IP packet for register is different (larger on the 3.8.6),
> but the important content of the Register message seems the same.
I've
> ruled out ISP/firewall interference, as its happened on a number of
> users.
> 
> Obviously there are fixes in 3.8.6, so I don't want to downgrade the
> phones again, but I can't see why they'd fail...
>  
> Adrian Marsh
>  
> 
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] asterisk configurator with E120P E1 card

[satish patel]

Dear all
   I want to configure 2 port E1 card on my asterisk so which 
version is best 1.2.x or 1.4.x can anyone suggest me which one is best right 
now for asterisk and anyone have configuration file to configure E1 card and 
zaptel.conf so i can configure it 





   
-
Boardwalk for $500? In 2007? Ha! 
Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Multiple servers using realtime

[Mindaugas Kezys]

That's a good note about MySQL replication. You can use it to remove
point-of-failure which currently is your DB server.

Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Collier
Sent: Thursday, August 23, 2007 12:21 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Multiple servers using realtime

I use a centralized database (with replication) for several servers, and it
works very well.  I keep all the mysql traffic on a separate network from
the SIP traffic. It makes it easy to add capacity.  If you are doing all the
mySQL on one box anyway, I don?t see any adavantage to using separate
databases.

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Peder @
NetworkOblivion
Enviado el: miercoles, 22 de agosto de 2007 19:06
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] Multiple servers using realtime


I am in the process of setting up several * servers using realtime and
connecting to mysql.  I am trying to figure out if I should just use one
database and one set of tables for all of the user data.  Or if I should
have separate databases for each * box.  The boxes are independent of
each other in that customerA only connects to box A.  They will never
fail over to box B or anything like that.  I want to use realtime for
queues,voicemail, sippeers and extensions.  The only issue that I have
come up with so far is that a common voicemail table would cause each
box to try and send out mwi indicators since it appears each * box pulls
all of the voicemail boxes from the DB every 10 seconds.

Any thoughts on whether I should go with one DB, or separate per box
DB's?  There is one mysql box, I am not referring to mysql on each box,
I am referring to whether I should use separate DB's within the one
mysql box for each * box.  Thanks.

Peder


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] B410P and echo

[Stefano Arata]

Hi, 

Where can I find some tools, such as ztmonitor for zaptel devices, to adjust
rxgain and txgain correctly on this card?
I've some troubles with finding the optimal configuration for the
echocancellator.


Thanks in advance,

Stefano Arata


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] contact header is missing in 200OK for SUBSCRIBE

[sumanth achar]

Hi,
 I am trying to SUBSCRIBE for message waiting indications to asterisk,
it sends 200 OK but contact header is missing(it is mandatory since
subscribe is dialog establishing method), due to which parsing fails, any
body knows about this issue...?

Regards,
Subramanya
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] ASTCC and IVR

[bilal ghayyad]

Hi list;

ASTCC supports IVR or there is a separate module for
IVR?

Can someone advise me a link to start download and
ready about ASTCC to do the configuration?

Regards,
-
ITS
IP Telephony and Contact Center Engineer
Bilal Ghayad
Mobile: 00865 9849460



   

Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for 
today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users