date:20060913


Moises Silva wrote:

So what should I do to build zaptel for the new kernel?


As Steve Totaro said, when running the newer kernel, go to zaptel 
sources and:


make clean
make && make install

I remembered that i had to "make linux26" && make install, but not
sure if this is still necessary for newer zaptel drivers.

"No such device or address" Is the error you get, so, i would bet you
dont have the file "/dev/zap/1" and "/proc/zaptel/" directory do you?
And if you dont have these, it means the zaptel drivers are not
loaded.

Try this:

modprobe zaptel
modprobe 

Regards



Dont forget ztcfg for good measure.
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RE: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls

2006-09-13 Thread Bill Gibbs

Make those calls then check the CLI "sip show channels" and see if the
channels are stay up

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Frederik
Fix
Sent: Wednesday, September 13, 2006 8:35 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working
togetherafter 3 calls

Hi,
I have a strange problem that I have no idea how to debug:

I have a Zyxel Prestige 2000W Wifi telephone that is connected to my  
Asterisk server which has a Junghanns.net QuadBRI card. I can make  
exactly 3 calls to the "outside" over the QuadBRI. Any calls after  
that fail with the log saying that all lines are busy.

Turning the phone off and on solves the problem and I can make 3  
calls again before it repeats. This problem does not occur when I  
make calls from my Cisco 7960G phones using SCCP or using eyebeam and  
SIP. Also making calls from the Zyxel through a cheap Cologne chipset  
ISDN card using zaphfc does not show this problem.

I am using the following versions:
Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r
Zyxel Prestige 2000W (version 1)
Zyxel-Firmware: Wj.00.11

Any help is very much appreciated,

Frederik

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Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS


Zeeshan Zakaria wrote:
Yes, I did rebuild zaptel after upgrading the OS. And ended up 
rebuilding everything, i.e. libpri, zaptel and asterisk, doing make 
clean on all of them. But still the problem persists. I have to load 
CentOS using the old kernel to keep the things working. As for 
spinlock, that error is in its version 4, but I am running ver 3.
 
So what should I do to build zaptel for the new kernel?


Boot to the new kernel and build zaptel.  Asterisk and libpri should be 
OK without rebuild.

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Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS

2006-09-13 Thread James Jones


Zeeshan Zakaria wrote:

Yes, I did rebuild zaptel after upgrading the OS. And ended up 
rebuilding everything, i.e. libpri, zaptel and asterisk, doing make 
clean on all of them. But still the problem persists. I have to load 
CentOS using the old kernel to keep the things working. As for 
spinlock, that error is in its version 4, but I am running ver 3.
 
So what should I do to build zaptel for the new kernel?




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I would try that.
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Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS

2006-09-13 Thread Moises Silva


So what should I do to build zaptel for the new kernel?


As Steve Totaro said, when running the newer kernel, go to zaptel sources and:

make clean
make && make install

I remembered that i had to "make linux26" && make install, but not
sure if this is still necessary for newer zaptel drivers.

"No such device or address" Is the error you get, so, i would bet you
dont have the file "/dev/zap/1" and "/proc/zaptel/" directory do you?
And if you dont have these, it means the zaptel drivers are not
loaded.

Try this:

modprobe zaptel
modprobe 

Regards

--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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RE: [asterisk-users] Polycom IP430 sound level too low?

2006-09-13 Thread David Gagnon

You must had the latest information in your sip.cfg. Compare the template
from version 1.6.7 and the one you are using.

David

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Louis-David
Mitterrand
Envoyé : 13 septembre 2006 05:55
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Polycom IP430 sound level too low?

Hello,

Has anyone noticed that the Polycom IP430 has a low incoming/outgoing 
sound level?

Is it a firmware issue or should I adjust my zap's tx/rxgain?
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Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS

2006-09-13 Thread Zeeshan Zakaria

Yes, I did rebuild zaptel after upgrading the OS. And ended up rebuilding everything, i.e. libpri, zaptel and asterisk, doing make clean on all of them. But still the problem persists. I have to load CentOS using the old kernel to keep the things working. As for spinlock, that error is in its version 4, but I am running ver 3.

 
So what should I do to build zaptel for the new kernel?
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[asterisk-users] QuadBRI and Zyxel Wifi phone stop working together after 3 calls

2006-09-13 Thread Frederik Fix


Hi,
I have a strange problem that I have no idea how to debug:

I have a Zyxel Prestige 2000W Wifi telephone that is connected to my  
Asterisk server which has a Junghanns.net QuadBRI card. I can make  
exactly 3 calls to the "outside" over the QuadBRI. Any calls after  
that fail with the log saying that all lines are busy.


Turning the phone off and on solves the problem and I can make 3  
calls again before it repeats. This problem does not occur when I  
make calls from my Cisco 7960G phones using SCCP or using eyebeam and  
SIP. Also making calls from the Zyxel through a cheap Cologne chipset  
ISDN card using zaphfc does not show this problem.


I am using the following versions:
Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r
Zyxel Prestige 2000W (version 1)
Zyxel-Firmware: Wj.00.11


Any help is very much appreciated,

Frederik

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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Steve Underwood


Steve Davies wrote:


On 9/13/06, Steve Underwood <[EMAIL PROTECTED]> wrote:


Artifex Maximus wrote:

If you look in http://www.soft-switch.org/download/snapshots/snapdsp,
the latest snapshot of spandsp and the app_rxfax and app_txfax
applications there provide ECM. It is less well tested than the
spandsp-0.0.2 code, but seems to be working pretty well now.



Fantastic.

This looks pretty good I have to say - The ECM seems as if it may be a
little intolerant... On a fax machine where I got 100% success in the
past with 0.0.2, I am now getting "result (60) Disconnected after
permitted retries." on about every 4th page.

Is the ECM tolerance level tuneable in spandsp, or is this
hard-defined in the standard? Is it just a matter of changing:
 #define MAX_MESSAGE_TRIES   3


Your problem probably has nothing to do with tolerance. If an exchange 
doesn't succeed after 3 tries, it is unlikely to ever succeed. You are 
probably hitting a bug. It is new code. :-)  Can you enable debug with 
"|debug" on the command line to rxfax/txfax, and send me the log?



I also noticed that the page title in the TIFF does not appear to be
set for the last page received any more (I have not looked into this
at-all though, so it may be my environment).


Do you mean the title line which spandsp can insert on each page? If so, 
I don't know how that can come out on any page with spandsp 0.0.2 or 
0.0.3. I fixed a bug in that last night. :-\ The next snapshot should 
have that fixed.


Steve

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Re: [asterisk-users] Polycom Firmware

2006-09-13 Thread BJ Weschke


On 9/13/06, Forum <[EMAIL PROTECTED]> wrote:

Unfortunately they pointed me back to Polycom and I have not yet heard back
from them.

Can somebody post a link to download sip2.0.1?



Your reseller should not be pointing you back to Polycom if they are
a certified reseller. The download is available to all certified
resellers via Polycom's extranet. However, there isn't a publicly
available link.

--
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http://www.btwtech.com/
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Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS

2006-09-13 Thread Steven Ringwald


Zeeshan Zakaria wrote:
Everything was working perfect until I updated CentOS using yum 
update. errors are like these. Please help, what is the solution for 
this. Obviously the zaptel hardware is not loading, byt why? How to 
load it again?
 
Sep 13 18:33:34 VERBOSE[1490] logger.c:  [chan_zap.so]Sep 13 18:33:34 
VERBOSE[1490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing 
'/etc/asterisk/zapata.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c:   
== Parsing '/etc/asterisk/zapata.conf': Found
Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing 
'/etc/asterisk/zapata-auto.conf': Sep 13 18:33:34 VERBOSE[1490] 
logger.c:   == Parsing '/etc/asterisk/zapata- auto.conf': Found
Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing 
'/etc/asterisk/zapata_additional.conf': Sep 13 18:33:34 VERBOSE[1490] 
logger.c:   == Parsing '/etc/asterisk/zapata_additional.conf': Found
Sep 13 18:33:34 WARNING[1490] chan_zap.c: Unable to specify channel 1: 
No such device or address
Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to open channel 1: No 
such device or address

here = 0, tmp->channel = 1, channel = 1
Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to register channel '1-23'
Sep 13 18:33:34 WARNING[1490] loader.c: chan_zap.so: load_module 
failed, returning -1

Sep 13 18:33:34 WARNING[1490] loader.c: Loading module chan_zap.so failed!



Did you rebuild the zaptel kernel drivers after upgrading the kernel???

Steve

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Re: [asterisk-users] Copyright issues with libcurl and OpenSSL

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

James Jones wrote:
> but in the register program it is staticly linked.

Maybe because you are top posting you didn't see my reply:

>
> Best place to ask this though is:
>
> http://licensing.digium.com/main_page.php

Some help if you want to post to the mailing lists:

1) Don't cross post (this means don't post to biz, dev and users at the
same time) - lot's of us are subscribed to all the lists and don't want
to have to read the same thing multiple times.

2) Don't top post.  There have been many discussions on this in the
past.  An example:

Yes I'm Sure

Really, are you sure?

Yeah it really does.

No it doesn't.

Because it makes it difficult to follow the conversation.

Why should I not top post?

3) Have a nice day!  :)

- --
Cheers,

Matt Riddell
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RE: [asterisk-users] Building Zaptel 1.2.9 with Octasic

2006-09-13 Thread shadowym

Say what?!

Asterisk Appliance on flash with GUI!

Looks pretty cool!  Blackfin RISC processor, 8 ports of FXO/FXS,
router+4port switch.  I've been waiting for someone to make something like
this!  The big question is what is it going to cost?
http://www.digium.com/en/docs/AADK001/AADK.pdf

 

-Original Message-
From: Steve Kennedy [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, September 13, 2006 11:19 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Building Zaptel 1.2.9 with Octasic

On Wed, Sep 13, 2006 at 01:34:30PM -0400, Mark Hulber wrote:

> Yes, it worked.  I didn't get the announcement of 1.2.9.1.

Seems it wasn't announced, nor Asterisk 1.2.12.1

Nor their new Asterisk Appliance that seems to run off Flash (with a GUI
that configures it all). ALso the new 4 port BRI card is on the site.


Steve

--
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro
Tech News Blog http://eurotechnews.blogspot.com


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Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS


Zeeshan Zakaria wrote:
Everything was working perfect until I updated CentOS using yum 
update. errors are like these. Please help, what is the solution for 
this. Obviously the zaptel hardware is not loading, byt why? How to 
load it again?
 
Sep 13 18:33:34 VERBOSE[1490] logger.c:  [chan_zap.so]Sep 13 18:33:34 
VERBOSE[1490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing 
'/etc/asterisk/zapata.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c:   
== Parsing '/etc/asterisk/zapata.conf': Found
Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing 
'/etc/asterisk/zapata-auto.conf': Sep 13 18:33:34 VERBOSE[1490] 
logger.c:   == Parsing '/etc/asterisk/zapata- auto.conf': Found
Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing 
'/etc/asterisk/zapata_additional.conf': Sep 13 18:33:34 VERBOSE[1490] 
logger.c:   == Parsing '/etc/asterisk/zapata_additional.conf': Found
Sep 13 18:33:34 WARNING[1490] chan_zap.c: Unable to specify channel 1: 
No such device or address
Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to open channel 1: No 
such device or address

here = 0, tmp->channel = 1, channel = 1
Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to register channel '1-23'
Sep 13 18:33:34 WARNING[1490] loader.c: chan_zap.so: load_module 
failed, returning -1

Sep 13 18:33:34 WARNING[1490] loader.c: Loading module chan_zap.so failed!

--
Zeeshan A Zakaria


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cd /usr/src/zaptel
make clean
make && make install
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Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Zeeshan Zakaria wrote:
> Everything was working perfect until I updated CentOS using yum update.
> errors are like these. Please help, what is the solution for this.
> Obviously
> the zaptel hardware is not loading, byt why? How to load it again?
> 
> Sep 13 18:33:34 VERBOSE[1490] logger.c:  [chan_zap.so]Sep 13 18:33:34
> VERBOSE[1490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
> Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing
> '/etc/asterisk/zapata.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c:   ==
> Parsing '/etc/asterisk/zapata.conf': Found
> Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing '/etc/asterisk/zapata-
> auto.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing
> '/etc/asterisk/zapata-auto.conf': Found
> Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing
> '/etc/asterisk/zapata_additional.conf': Sep 13 18:33:34 VERBOSE[1490]
> logger.c:   == Parsing '/etc/asterisk/zapata_additional.conf': Found
> Sep 13 18:33:34 WARNING[1490] chan_zap.c: Unable to specify channel 1: No
> such device or address
> Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to open channel 1: No such
> device or address
> here = 0, tmp->channel = 1, channel = 1
> Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to register channel '1-23'
> Sep 13 18:33:34 WARNING[1490] loader.c: chan_zap.so: load_module failed,
> returning -1
> Sep 13 18:33:34 WARNING[1490] loader.c: Loading module chan_zap.so failed!

New kernel maybe?

Zaptel kernel modules compiled for older kernel?

Try recompiling zaptel against new one?

Also search for CentOS Zaptel bug in Google.  Something about spinlock.h
or similar if I remember correctly.

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] Copyright issues with libcurl and OpenSSL

2006-09-13 Thread James Jones


but in the register program it is staticly linked.

Matt Riddell (IT) wrote:


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

James Jones wrote:
 


Does anyone know why the g729 codec module sold by diguim  does not
display the OpenSSL copyright information. Do they have an agreement
with OpenSSL to not display the Copyright Information that is required
ny their license when distributed as part of a binary that uses OpenSSL.

The registration program uses libcurl and openssl (both statically
linked) to register the g.729 codec (and they said soon other
products).  This too does not display the required information about
other peoples open source code.
   



Don't know about libcurl, but have you read the LICENCE file in the
Asterisk directory?

"Specific permission is also granted to link Asterisk with OpenSSL and
OpenH323."

I assume if they have specific permission, then they've sorted something
with the OpenSSL people.

Best place to ask this though is:

http://licensing.digium.com/main_page.php

- --
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Matt Riddell
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Re: [asterisk-users] Via Epia platforms and asterisk

2006-09-13 Thread Kristian Kielhofner


Chris Bagnall wrote:

Greetings list,

Has anyone done any research into call routing and transcoding performance
using a Via Epia based platform?

We have a client with a box in a datacentre with 2 PRIs going into the
machine. We've moved most of their asterisk handling into another more cost
effective datacentre, but the cost of moving the PRIs is prohibitive.

(to give you an idea, they're paying as much for the single rackspace in the
first DC as they are for a whole rack in the new DC)

In a nutshell, what we're looking to do is to replace the asterisk box in
the expensive datacentre with a low power consumption 1U chassis that'll
simply take calls coming in on the PRIs and kick them out via IAX to one or
other of the new asterisk servers in the other datacentre.

In an ideal world, the first box would also transcode the calls to something
like g729 or (ideally) Speex to save bandwidth between the two locations
(since the DC also charges for that), but I'm not sure that's realistic
given the hardware we're looking to use.

Any thoughts gratefully appreciated.

Regards,

Chris


Chris,

	While I have not done extensive testing, Asterisk seems to be quite 
capable on VIA hardware.  I like VIA hardware so much that I compile a 
special AstLinux image for them...


	Why don't you just keep is ulaw and IAX2 trunk in between sites?   That 
would use at most about ~6000kbps with both PRIs at %100. The VIAs could 
definitely handle that.


--
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[asterisk-users] Asterisk as a B2BUA and/or a FXS-SIP gateway?

2006-09-13 Thread AK 4asterisk

Hi All,This is my first email on this list so not sure what the protocol out here is. Anyways, I am trying to configure Asterisk in such a way that it would act more like a B2BUA rather than as a PBX. My customers get SIP service from their service providers and have planned on installing SIP soft phones. I was wondering whether one could configure Asterisk such that all requests coming in from the SIP phones on the LAN could be routed to an Application Server in the SP's network. Of course some intelligence would reside in the Asterisk but it would use the softswitch as its Proxy/Registrar/Feature server. In case connection to the SP goes down, then Asterisk would take over to provide basic functionality. Also using some FXS and FXO cards some kind of a fall back could be provided. Has anyone tried this out?
- AK
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Re: [asterisk-users] University switches to Asterisk

2006-09-13 Thread broadbandvoice


Thats good news for us.
 
-- Original message -- From: Doug Lytle <[EMAIL PROTECTED]> > Interesting article I found linked from Groklaw: > > "Sam Houston State University replaces Cisco CallManagers, Nortel PBXs > with Linux-based VoIP and messaging servers" > > http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1 > > 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 Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo
 /aster
isk-users 

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[asterisk-users] chan_zap.so stopped working after upgrading CentOS

2006-09-13 Thread Zeeshan Zakaria

Everything was working perfect until I updated CentOS using yum update. errors are like these. Please help, what is the solution for this. Obviously the zaptel hardware is not loading, byt why? How to load it again?

 
Sep 13 18:33:34 VERBOSE[1490] logger.c:  [chan_zap.so]Sep 13 18:33:34 VERBOSE[1490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing '/etc/asterisk/zapata.conf': Sep 13 18:33:34 VERBOSE[1490] 
logger.c:   == Parsing '/etc/asterisk/zapata.conf': FoundSep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing '/etc/asterisk/zapata-auto.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing '/etc/asterisk/zapata-
auto.conf': FoundSep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing '/etc/asterisk/zapata_additional.conf': Sep 13 18:33:34 VERBOSE[1490] logger.c:   == Parsing '/etc/asterisk/zapata_additional.conf': FoundSep 13 18:33:34 WARNING[1490] chan_zap.c: Unable to specify channel 1: No such device or address
Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to open channel 1: No such device or addresshere = 0, tmp->channel = 1, channel = 1Sep 13 18:33:34 ERROR[1490] chan_zap.c: Unable to register channel '1-23'
Sep 13 18:33:34 WARNING[1490] loader.c: chan_zap.so: load_module failed, returning -1Sep 13 18:33:34 WARNING[1490] loader.c: Loading module chan_zap.so failed!-- Zeeshan A Zakaria 
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Re: [asterisk-users] Copyright issues with libcurl and OpenSSL

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

James Jones wrote:
> Does anyone know why the g729 codec module sold by diguim  does not
> display the OpenSSL copyright information. Do they have an agreement
> with OpenSSL to not display the Copyright Information that is required
> ny their license when distributed as part of a binary that uses OpenSSL.
> 
> The registration program uses libcurl and openssl (both statically
> linked) to register the g.729 codec (and they said soon other
> products).  This too does not display the required information about
> other peoples open source code.

Don't know about libcurl, but have you read the LICENCE file in the
Asterisk directory?

"Specific permission is also granted to link Asterisk with OpenSSL and
OpenH323."

I assume if they have specific permission, then they've sorted something
with the OpenSSL people.

Best place to ask this though is:

http://licensing.digium.com/main_page.php

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] OT -- echo cancellation of an audio file

-BEGIN PGP SIGNED MESSAGE-
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William M Conlon wrote:
> I recorded an internet radio program using iTunes, and somehow got an echo.
> 
> Anyone have any suggestions on how to remove echo from an existing file?
> 
> My wife was on the program, and I promised to record it for her, so I'm
> in the hot seat :).

Er, don't know if this will work well, but:

1) Convert to wav, mono
2) Measure the delay in the echo (find a peak and select to the
repetition of the same peak)
3) Measure the difference in volume between the original peak and the
repeated peak.
4) Create a new file with a silence at the beginning of the length of
the measured delay (in 2), followed by the original sound
5) Lower the volume of the new file by the amount found in 3 (don't
forget the peaks need to stand alone so you get an accurate reading)
6) Invert the phase of the new file
7) Put the new quieter, delayed and inverted copy of the file into the
left channel of a new stereo audio file
8) Put the original into the right
9) Export the file as a new mono file by summing and /2

The result should be reasonably reduced echo.

If this doesn't make sense or you need help with it, mail me, and I'll
have a go here.

- --
Cheers,

Matt Riddell
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[asterisk-users] Copyright issues with libcurl and OpenSSL

2006-09-13 Thread James Jones

Does anyone know why the g729 codec module sold by diguim  does not 
display the OpenSSL copyright information. Do they have an agreement 
with OpenSSL to not display the Copyright Information that is required 
ny their license when distributed as part of a binary that uses OpenSSL.


The registration program uses libcurl and openssl (both statically 
linked) to register the g.729 codec (and they said soon other 
products).  This too does not display the required information about 
other peoples open source code.


If they don't have already signed agreements with the developer of 
libCurl and OpenSSL they are distrabuting libCURL and OpenSSL with 
giving the correct copyright information with register program and then 
they are making money off the OpenSSL libs with they sell the g729 codec 
with out giving correct copyright information. I just want to confirm 
they have agreement not to show copyright information for the products, 
if not it could cause legal issues for people who work with Asterisk 
when they distribute it to their clients/customers.


I have enclosed relivent snippites of the licenses in question and links 
to the full licenses.


libCurl: http://curl.haxx.se/docs/copyright.html

COPYRIGHT AND PERMISSION NOTICE

Copyright (c) 1996 - 2006, Daniel Stenberg, <[EMAIL PROTECTED]>.

All rights reserved.

Permission to use, copy, modify, and distribute this software for any purpose
with or without fee is hereby granted, provided that the above copyright
notice and this permission notice appear in all copies.

THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OF THIRD PARTY RIGHTS. IN
NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE
OR OTHER DEALINGS IN THE SOFTWARE.

Except as contained in this notice, the name of a copyright holder shall not
be used in advertising or otherwise to promote the sale, use or other dealings
in this Software without prior written authorization of the copyright holder.

OpenSSL: http://www.openssl.org/source/license.html

/* 
* Copyright (c) 1998-2006 The OpenSSL Project.  All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
*notice, this list of conditions and the following disclaimer.
*
* 2. Redistributions in binary form must reproduce the above copyright
*notice, this list of conditions and the following disclaimer in
*the documentation and/or other materials provided with the
*distribution.
*
* 3. All advertising materials mentioning features or use of this
*software must display the following acknowledgment:
*"This product includes software developed by the OpenSSL Project
*for use in the OpenSSL Toolkit. (http://www.openssl.org/)"
*
* 4. The names "OpenSSL Toolkit" and "OpenSSL Project" must not be used to
*endorse or promote products derived from this software without
*prior written permission. For written permission, please contact
*[EMAIL PROTECTED]
*
* 5. Products derived from this software may not be called "OpenSSL"
*nor may "OpenSSL" appear in their names without prior written
*permission of the OpenSSL Project.
*
* 6. Redistributions of any form whatsoever must retain the following
*acknowledgment:
*"This product includes software developed by the OpenSSL Project
*for use in the OpenSSL Toolkit (http://www.openssl.org/)"


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Re: [asterisk-users] Polycom related question

2006-09-13 Thread Kevin Smith


John Marvin wrote:

Kevin Smith wrote:

Here is what the configuration looks like for one of the phones, the 
other is 284:


[283](Empire-Defaults)
[EMAIL PROTECTED]

[283a](Empire-Defaults)   [EMAIL PROTECTED]

[283b](Empire-Defaults)
[EMAIL PROTECTED]



So actually you are trying to use one phone to monitor (receive 
notifies for) multiple boxes. It looks like the Polycom's have some 
support a different mwi for each registration, but I'm not sure how 
well it works. 

Right. It sort of breaks down like this: (I pray this keeps the formatting)

Phone 283   Phone 284
Line 1: VM box 283Line 1: VM box 283
Line 2: VM box 284Line 2: VM box 284
Line 3: VM box 285Line 3: VM box 285

So really one line on the phone is just looking at one mail box, but 
there are two phones per mail box.
You didn't specify what username you specify for each config above, so 
I don't know if the notifies are going to one registration or to 
different registrations. The messages button on the phone only seems 
to show the status of one registration, but the indicator light seems 
to combine the different results together (and you can't clear the 
light with the clear button since that only applies to one of the 
registrations). Of course that assumes that you are sending the 
notifies to different registrations on the phone -- all bets are off 
if you are sending them to the same registration (which is controlled 
by the username value) since Asterisk is treating them as separate 
phones the notifies will collide with eachother.
Well the phone will take the sip extension as the username. And I pass 
the username of the voice mail box in the to the voice mail function 
depending on which line it is calling from in the dialplan. That is if I 
am following your statement about which username I specify correctly. If 
I am not, then this probably won't make much sense.


You would get more reliable behaviour if you did as Rich suggested and 
just specified something like this for just the [283] config:


[EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED]

In that case Asterisk sums up the total messages in each of the boxes 
and the messages button on the phone will show you that total rather 
than the results for only one of the boxes.

I'll give it a try the next time I am in the office and see what happens.


The polycom documentation is not very clear on how multiple mwi's are 
supposed to work, so I'm not sure what the right answer is.

AGREED!


John
Thanks again with the suggestions, I'll let you know the results when I 
get a chance to try it out.


Kevin
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[asterisk-users] Via Epia platforms and asterisk

2006-09-13 Thread Chris Bagnall

Greetings list,

Has anyone done any research into call routing and transcoding performance
using a Via Epia based platform?

We have a client with a box in a datacentre with 2 PRIs going into the
machine. We've moved most of their asterisk handling into another more cost
effective datacentre, but the cost of moving the PRIs is prohibitive.

(to give you an idea, they're paying as much for the single rackspace in the
first DC as they are for a whole rack in the new DC)

In a nutshell, what we're looking to do is to replace the asterisk box in
the expensive datacentre with a low power consumption 1U chassis that'll
simply take calls coming in on the PRIs and kick them out via IAX to one or
other of the new asterisk servers in the other datacentre.

In an ideal world, the first box would also transcode the calls to something
like g729 or (ideally) Speex to save bandwidth between the two locations
(since the DC also charges for that), but I'm not sure that's realistic
given the hardware we're looking to use.

Any thoughts gratefully appreciated.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Artifex Maximus


Hello Steve,

Thank you! I definitely try tomorrow but now it's time to sleep.

bye,
Zsolt

On 9/13/06, Steve Underwood <[EMAIL PROTECTED]> wrote:

Hi Bruce,

Looks like your typing is as bad as mine :-)

Try http://www.soft-switch.org/downloads/snapshots/spandsp

Steve

Bruce Reeves wrote:

> Try, http://www.soft-switch.org/downloads/snapshots/snapdsp
> ,
>
> On 9/13/06, *Artifex Maximus* <[EMAIL PROTECTED]
> > wrote:
>
> Hello Steve,
>
> On 9/13/06, Steve Underwood <[EMAIL PROTECTED]
> > wrote:
> > Artifex Maximus wrote:
> >
> > > Hello,
> > >
> > > I had received a lot of unreadable pages with rxfax. I've been
> doing
> > > some search on net and found this:
> > > http://threebit.net/mail-archive/asterisk-users/msg15708.html
> > >
> > > It looks like rxfax/spandsp doesn't support ecm error
> correction. Bad
> > > news for me. Is it still the case? app_rxfax.c dated as 8th of
> > > february so I think the answer is yes but I am still hoping a
> little
> > > no or might someone have a patch for enabling/implementing ecm.
> >
> > If you look in
> http://www.soft-switch.org/download/snapshots/snapdsp,
> > the latest snapshot of spandsp and the app_rxfax and app_txfax
> > applications there provide ECM. It is less well tested than the
> > spandsp-0.0.2 code, but seems to be working pretty well now.
>
> Sounds promising but gives me
> "Not Found
>
> The requested URL /download/snapshots/snapdsp was not found on
> this server.
> Apache/2.0.52 (CentOS) Server at www.soft-switch.org
>  Port 80"

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[asterisk-users] Polycom MyStat

2006-09-13 Thread Douglas Garstang

Has anyone ever gotten the Polycom MyStat soft-key to do anything?

Setting the status to something like 'Away', does not generate any outgoing SIP 
traffic from the phone. Calling into the phone either from a watched buddy, or 
other number, acts as if the status was never changed. A call to Polycom 
yielded no results. 

Thanks,
Doug.
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Re: [asterisk-users] Snom 360 Function Keys

2006-09-13 Thread Conrad Wood


> 2) One digium quadri primary ISDN interface (TE410P)
> 3) Two Rhino Channel Banks
> 4) 25 Analogue Phones on every channel bank
> 
> How I can configure function keys on my SNOM 360 for monitoring analogue 
> phone status?

I haven't used the Rhino Channel banks yet, so I'm guessing to some
degree here:
I'm not exactly sure how you address each phone on the channel bank.
Presumably it connects to the digium card. If so, don't you have
something like ZAP/1 to dial first phone ZAP/2 to dial second etc?

If so, you should be able to add hints to your dialplan for each phone
and make the snom monitor those.
The snom360 works rather well with hinting and allows you to
call/transfer a call to the monitored phone when you press the button
too.

For example...

in the dialplan:
exten => 4101,hint,Zap/1

for the functionkey (type destination) put:



Conrad.

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Re: [asterisk-users] Polycom 501 display bug

2006-09-13 Thread Jim Rice

On Wed, 2006-09-13 at 13:10 -0700, William M Conlon wrote:
> I'm running SIP version 1.6.2.0041 on a Polycom 501.
> 
> The display truncates the date today, showing:
> 
> Wednesday, September,...
> 
> I would prefer it either line break and finish, or abbreviate when  
> the date won't display completely
> 
> Anyone know if this is patched in later versions of the SIP  
> applications?
> 
> Bill

It's a Preference Setting on the phone itself.


Settings...
   Basic...
  1. Preferences
 2. Time & Date
1. Clock Date

>From there you have a choice of 14 different displays.

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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Rosario Pingaro


I am testing the last snapshot.
It seems to work now with a lot of fax machine that was not working 
before.


But I have a very strange problem:
the fax2mail script that is present after rx_fax is not been executed 
now...


Very very strange
Any idea?

Regards
Rosario

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


Sent: Wednesday, September 13, 2006 11:21 AM
Subject: Re: [asterisk-users] rxfax, spandsp and lack of ecm



Artifex Maximus wrote:


Hello,

I had received a lot of unreadable pages with rxfax. I've been doing
some search on net and found this:
http://threebit.net/mail-archive/asterisk-users/msg15708.html

It looks like rxfax/spandsp doesn't support ecm error correction. Bad
news for me. Is it still the case? app_rxfax.c dated as 8th of
february so I think the answer is yes but I am still hoping a little
no or might someone have a patch for enabling/implementing ecm.


If you look in http://www.soft-switch.org/download/snapshots/snapdsp, the 
latest snapshot of spandsp and the app_rxfax and app_txfax applications 
there provide ECM. It is less well tested than the spandsp-0.0.2 code, but 
seems to be working pretty well now.


Steve

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Re: [asterisk-users] OT -- echo cancellation of an audio file

2006-09-13 Thread Jean-Michel Hiver


William M Conlon a écrit :

I recorded an internet radio program using iTunes, and somehow got an  
echo.


Anyone have any suggestions on how to remove echo from an existing file?


Convert it to gsm, send it through asterisk (by calling yourself), 
activate echo canceller, and record the call? Note that you might end up 
with crappier audio doing that... :)


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[asterisk-users] OT -- echo cancellation of an audio file

2006-09-13 Thread William M Conlon

I recorded an internet radio program using iTunes, and somehow got an  
echo.


Anyone have any suggestions on how to remove echo from an existing file?

My wife was on the program, and I promised to record it for her, so  
I'm in the hot seat :).


Bill


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Re: [asterisk-users] Queue - static members

2006-09-13 Thread Kevin Bockman


Tomislav Parčina wrote:

I have queue with member defined as:
member => Agent/SIP/148,1
member => Agent/SIP/143,2 


And when I do show queues this is what I see.

pbx*CLI> show queues
prodaja  has 0 calls (max 5) in 'rrmemory' strategy (0s holdtime), W:10, C:0
, A:1, SL:0.0% within 60s
   Members:
  Agent/SIP/148 with penalty 1 (Unavailable) has taken no calls yet
  Agent/SIP/143 with penalty 2 (Unavailable) has taken no calls yet


You probably already figured this out, but you use either Agent or SIP, 
not both.  Use Agent if they login through AgentLogin or SIP if it is 
calling the SIP phone directly.



Kevin
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RE: [asterisk-users] Polycom Firmware

2006-09-13 Thread Forum

Unfortunately they pointed me back to Polycom and I have not yet heard back
from them. 

Can somebody post a link to download sip2.0.1?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of stoffell
Sent: Wednesday, September 13, 2006 4:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Firmware

On 9/13/06, Forum Expansive <[EMAIL PROTECTED]> wrote:
>
> What is the latest polycom firmware and where can I get it?

sip2.0.1, ask your reseller, they must give it to you.

cheers
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[asterisk-users] Polycom 501 display bug

2006-09-13 Thread William M Conlon


I'm running SIP version 1.6.2.0041 on a Polycom 501.

The display truncates the date today, showing:

Wednesday, September,...

I would prefer it either line break and finish, or abbreviate when  
the date won't display completely


Anyone know if this is patched in later versions of the SIP  
applications?


Bill
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[asterisk-users] Jitter Buffer on SIP

2006-09-13 Thread ggonzalez

Hello all! 
I dont know how affect this issue (jitter buffer) on a SIP implementation with
a
VOIP trunk and I want to know how to setup this item to get a good IP quality
calls without voice delay. thanks for any help.





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[asterisk-users] Astmanproxy authentication problems

2006-09-13 Thread fho

Hello,

I've try to use Astmanproxy with "Asterisk TAPI line".
But login fails,  astmanproxys error message:

"Sep 13 20:06:26: [EMAIL PROTECTED] got: Response: Error
Sep 13 20:06:26: [EMAIL PROTECTED] got: Message: No variable specified
Sep 13 20:06:26: [EMAIL PROTECTED]: attempting read..."

AStmanproxy has successfull connected to astmanager.

Asterisk TAPI Line also can successfull connect to the Asterisk manager,
but when i choose Astmanproxy+correct Port (1234) I get "login failed"...

I've configured astmanproxy, to require authentication, with password
foobar... I've also tried to use seperate users from
astmanproxy.users... But authentication fails, too...

Here is my astmanproxy.conf:

"host = localhost, 5038, admin, amp111, on, off
retryinterval = 2
maxretries = 10
sslclienthellotimeout = 200
acceptencryptedconnection = yes
acceptunencryptedconnection = yes
asteriskwritetimeout=100
clientwritetimeout=200
certfile = /var/lib/asterisk/certs/proxy-server.pem
listenaddress = *
listenport = 1234
authrequired = yes
proxykey = foobar
proc_user = nobody
proc_group = nobody
inputformat = standard
outputformat = standard
autofilter = off
logfile = /var/log/asterisk/astmanproxy.log"

I use AStmanproxy ver. 1.21.


What can be wrong?


thanks

greetings

Fabian



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Re: [asterisk-users] set global variable

2006-09-13 Thread Jan Fousek

Thanks for ideas, I thing I've found another way. It's possible to store data 
in 
database, alter them via manager and use in dialplan. Exactly, what I needed.
Just hope,  it will work

Jan
__
> Od: [EMAIL PROTECTED]
> Komu: Asterisk Users Mailing List - Non-Commercial 
> Discussion
> Datum: 13.09.2006 21:45
> Předmět: Re: [asterisk-users] set global variable
>
>Jan Fousek wrote:
>> Hi all,
>>  is there any possibility of setting the global variables from outside
of asterisk?
>> Like manager api or something like that.
>>
>> Thanks a lot
>>
>>   not sure about current svn trunk,
>
>but in the past you could set a channel var with
>
>action: SetVar
>channel: Zap/49-1
>variable: SOMEVAR=SOMESTRING
>
>but required a channel
>
>action: GetVar will get channel or global vars
>
>you should be able to easily mod manager.c to use ast_get_channel_by_name
>code if channel is passed, and use pbx_builtin_setglobalvar otherwise.
>
>sad thing is setvar is depeciated in favor of set.  (reminder, no clue
what current svn trunk has)
>
>i hope this helps
>
>
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Re: [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya

BJ Weschke wrote:

On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

BJ Weschke wrote:
> On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> Doug Lytle wrote:
>> > Steve Totaro wrote:
>> >> I am trying to connect a Definity G3 to an asterisk system. I 
had it

>> >> working OK with the exception of the caller ID on the Definity
>> >> handsets just
>> >
>> > He wants to know if your Definity is an S, SI or an R?
>> >
>> > Doug
>> >
>> I am not sure, it is a refrigerator sized unit with three 
cabinets. The
>> manual was printed in 1988 and is ATT&T Definity Communications 
System

>> Generic1 and Generic3.
>>
>> If you need more, I can go to the remote system and provide more
>> details.
>>
>> I REALLY appreciate you helping on this.
>>
>
> Sounds like a G3R. How are you signaling between the two? PRI?
>
Yes. I have two ISDN communication processor links setup which I later
read on google was not supported. Now I cannot remove the new entry. I
just get "identifier not assigned"

What you'll want to do in order to set this up, is build a trunk
group in DSA that consists of the two T1's, and then you'll want to
build a signaling group that uses "isdn" signaling and then (probably
3-4 pages over in DSA) you'll want to assign all the channels of those
T1s as "members" of that signaling group.

On the asterisk side you'll want to setup the trunks as pri_net.

Put a T1 crossover cable in between, if there's no carrier loops
involved, and then you're done.

OK,

Here is what I did.

I setup a ds1 with standard T1 pri settings and set to the network.

I setup a trunk-group with all the ports on that ds1, set it to isdn-pri 
and tie and set the signaling to the step below

I setup a signaling-group with the primary d-channel as the 24rth port 
on the ds1 using associated signaling

I setup a data-module

I setup communication-interface link

I setup communication-interface processor to isdn h priority

I went back to communication-interface link and enabled it.

Everything worked perfectly. The d channel came up and 23 b channels 
through the definity and the new ivr server. Calls in and out sounded 
great and no PRI errors whatsoever.

I repeated the same steps above with the same settings but obviously 
different boards and identifiers.

Now the D channel goes up and down on both links and the b channels 
never come up

I remove everything I just added (new trunk-group, signaling group)

I tried to remove the data-module but it errors saying I have to remove 
it from the communication-interface link

I tried to remove the data-module from the communication-interface link 
but that errors saying I have to remove the communication-interface 
processor

I tried to blank out the second entry for the communication-interface 
processor and it errors saying “Identifier not Assigned”

I have busied out everything related to the board and data-module and 
also tried to change the second isdn value in communication-interface 
processor but once I get all the values filled in, I get the same error 
“Identifier not Assigned”

The original PRI still does not work. The D channel comes up and down 
and no B channels.

Thanks,
Steve

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Re: [asterisk-users] set global variable

2006-09-13 Thread Richard Lyman


Jan Fousek wrote:

Hi all,
 is there any possibility of setting the global variables from outside of 
asterisk?
Like manager api or something like that.

Thanks a lot

  

not sure about current svn trunk,

but in the past you could set a channel var with

action: SetVar
channel: Zap/49-1
variable: SOMEVAR=SOMESTRING

but required a channel

action: GetVar will get channel or global vars

you should be able to easily mod manager.c to use ast_get_channel_by_name
code if channel is passed, and use pbx_builtin_setglobalvar otherwise.

sad thing is setvar is depeciated in favor of set.  (reminder, no clue 
what current svn trunk has)


i hope this helps


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Re: [asterisk-users] Building Zaptel 1.2.9 with Octasic

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mark Hulber wrote:
> Yes, it worked.  I didn't get the announcement of 1.2.9.1.

[Shameless self plug]

Read the Daily Asterisk News, it was announced there with changes today.

:)

[/Shameless self plug]

- --
Cheers,

Matt Riddell
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Re: [asterisk-users] Asterisk crashing when monitoring SIP device with Chanspy

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

[EMAIL PROTECTED] wrote:
> Forgive me if this thread already exists -
>  
> Am running asterisk 1.2.6.1 - Every so often, asterisk crashes when doing a 
> Chanspy on a SIP phone monitoring a sip phone.
>  
> Dialplan is set up as follows:
>  
> exten => 6274,1,ChanSpy(SIP/403274)
>  
> Intra-building we dial 3 digit extensions, monitor is obviously 6+3 digit 
> extension
>  
> I don't think it's a utilization issue, it seems to happened when there were 
> 80 phones on the network, and still happens with 160 phones. I can tell when 
> it happens, a manager will call me and say they have 15 calls in queue but no 
> calls are getting to the agents, then a minute later I get a call complaining 
> about 1 way audio, then the asterisk daemon stops - I have to start asterisk.
>  
> There is nothing in the log file either - except the last entry being NOTICE 
> - ChanSpy attaching to SIP/blahblah then no entries after.
>  
> I saw in bug tracker 3851 about people having crashing issues but that was 
> over a year ago.
>  
> Anyone else having problems or enlighten me on a fix?

Upgrade.

There have been some significant changes in the code related to spies
being applied to channels.

These have also been committed to the stable version (I.E. 1.2 branch)

- --
Cheers,

Matt Riddell
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[asterisk-users] Unknown RTP codec 100 received

2006-09-13 Thread Thomas Artner

Hi!

I have an analog fax machine connected to a sipura ATA which is connected to 
my asterisk box.
In my asterisk box I have a digium card for a connection to the public 
telephone network (analog).
On this digium card is also a us robotics sportster modem (analog) connected.

My problem:
I can send faxes from the analog faxmachine to the us robotics modem, but when 
I try to send a fax to someone else (via the public phone network) it fails 
and I get the following error message: Unknown RTP codec 100 received.

Does anyone have an idea why faxing to the analog modem works perfectly, and 
why it fails if i try to send one "outside" ?

thx in advance,
tom
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[asterisk-users] success

2006-09-13 Thread Christopher Corn

i finished setting my asterisk pbx with 5 phones. thanks for everyones help here, in getting this accomplished. it is greatly appreciated. this is what i set up. Athlon 3500+ cpu (2ghz i think) 1 Gig of RAM Netopia gateway to SBC DSL 6000kbps/600kbps Linksys wire with QOS functionality Trixbox, latest version, 1.1? origination and termination, voip service provided by VOIPSTREET Grandstream 100 phones g729 codec I make concurrent phone calls out to popcorn (time service) and the codec worked just like it said it would, i counted 10kbps for each phone with linux, maybe extra over head not related to the codec. voipstreet is also very quick to respond to trouble tickets and their Asterisk procedure if very clear and concise. so far call quality is excellent even with concurrent calls.
total cost was very low. i used a system laying around so that was most of the cost Grandstream phones - 55$ each Linksys router with QOS, WRT54G - 60$ VOIP service number porting - 15$ 45$ - testing out a voip service and not liking the service and not getting a refund, axvoice.com. apparently they don't give refunds on byod (bring your own device, i.e.; asterisk) setups again, thanks all, just thought i'd post this here, so that other people may be able to possibly find this info useful.___
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[asterisk-users] Asterisk crashing when monitoring SIP device with Chanspy

2006-09-13 Thread mezzmor


Forgive me if this thread already exists -


 


Am running asterisk 1.2.6.1 - Every so often, asterisk crashes when doing a Chanspy on a SIP phone monitoring a sip phone.


 


Dialplan is set up as follows:


 


exten => 6274,1,ChanSpy(SIP/403274)


 


Intra-building we dial 3 digit extensions, monitor is obviously 6+3 digit extension


 


I don't think it's a utilization issue, it seems to happened when there were 80 phones on the network, and still happens with 160 phones. I can tell when it happens, a manager will call me and say they have 15 calls in queue but no calls are getting to the agents, then a minute later I get a call complaining about 1 way audio, then the asterisk daemon stops - I have to start asterisk.


 


There is nothing in the log file either - except the last entry being NOTICE - ChanSpy attaching to SIP/blahblah then no entries after.


 


I saw in bug tracker 3851 about people having crashing issues but that was over a year ago.


 


Anyone else having problems or enlighten me on a fix?


 


Thanks


 


 



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Re: [asterisk-users] Dropped Calls on TDM400p

2006-09-13 Thread Jonathan Barratt

And of course I find the HCL right after clicking send. 

My MB is not listed as having any known issues.  I still haven't
found an approved list, so I can't say that it's approved
either.   But if swapping the TDM400P's PCI slot doesn't fix
the problem I am down to replace the MB and do a full re-install
(again).

Gratefully,
JonathanOn 9/13/06, Jonathan Barratt <[EMAIL PROTECTED]> wrote:
Thanks for the reply Steve.

I am calm now.  :)

I've been getting the exact time and number of the dropped calls for
the last couple weeks, and there was nothing in system or asterisk logs
at those times.  So I've spent the last three days sitting at the
server, in their office.  I can see nothing out of the ordinary
going on when the calls are dropped.  It seems to happen mostly on
outbound calls.
The TDM400p is on the same interrupt as the graphics card but nothing else.  I didn't think this 
would be an issue as the box was running headless.  So I guess first up is changing PCI slots.
MB is an EPoX EP-8KTA2L.  I haven't yet found the Hardware
Compatibility List on digium.com to determine if it's supported or has
known issues, but will keep looking.

I was expecting issues, and we've had many of them (esp. echo), but
I've been able to resolve them all myself with research and
experimentation, except for this persistent intermittent dropped call
problem...

I'm really grateful for your input Steve, please keep it coming!

Thanks very much!
Jonathan

On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

Sorry, see now that it is pots.How do your interrupts look?  What is the hardware platform or morespecifically the MB?  Is the platform listed on Digium's site asapproved or listed as having issues?

Thanks,SteveSteven Totaro wrote:>> They need to document the exact day and time so you can look in the> logs.  Is this a T1 or POTS? Customers always complain and threaten to go back to their old PBX.
> First, calm down, then calm them down and make sure they know you are> working on it.  Every new install is going to have issues that will> take time to resolve.  Remember that in your pricing or you will soon
> be out of business. Get exact times and frequency then check your logs to see if anything> matches that may be an issue. Thanks,>
> Steve
 >> *From:* 
[EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]] *On Behalf Of> *Jonathan Barratt
> *Sent:* Wednesday, September 13, 2006 1:02 PM> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Dropped Calls on TDM400p These are just PSTN calls, and I have set busydetect=no and> callprogress=no in zapata.conf as per voip-info guidance, but problem
> persists.>> CPU load never breaks 20, so that doesn't seem to be the problem, but> it's a 1.2Ghz Athlon with 768MB RAM.>> Power supply to system is clean, there's no heavy network traffic
> going on besides Asterisk to the phones (Aastra 480i's).>> What other factors can I investigate?>> This client is so unhappy they are ready to go back to their old PBX> system.

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Re: [asterisk-users] Dropped Calls on TDM400p

2006-09-13 Thread Jonathan Barratt

Thanks for the reply Steve.

I am calm now.  :)

I've been getting the exact time and number of the dropped calls for
the last couple weeks, and there was nothing in system or asterisk logs
at those times.  So I've spent the last three days sitting at the
server, in their office.  I can see nothing out of the ordinary
going on when the calls are dropped.  It seems to happen mostly on
outbound calls.
The TDM400p is on the same interrupt as the graphics card but nothing else.  I didn't think this 
would be an issue as the box was running headless.  So I guess first up is changing PCI slots.
MB is an EPoX EP-8KTA2L.  I haven't yet found the Hardware
Compatibility List on digium.com to determine if it's supported or has
known issues, but will keep looking.

I was expecting issues, and we've had many of them (esp. echo), but
I've been able to resolve them all myself with research and
experimentation, except for this persistent intermittent dropped call
problem...

I'm really grateful for your input Steve, please keep it coming!

Thanks very much!
Jonathan

On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
Sorry, see now that it is pots.How do your interrupts look?  What is the hardware platform or morespecifically the MB?  Is the platform listed on Digium's site asapproved or listed as having issues?
Thanks,SteveSteven Totaro wrote:>> They need to document the exact day and time so you can look in the> logs.  Is this a T1 or POTS? Customers always complain and threaten to go back to their old PBX.
> First, calm down, then calm them down and make sure they know you are> working on it.  Every new install is going to have issues that will> take time to resolve.  Remember that in your pricing or you will soon
> be out of business. Get exact times and frequency then check your logs to see if anything> matches that may be an issue. Thanks,>> Steve
 >> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]] *On Behalf Of> *Jonathan Barratt> *Sent:* Wednesday, September 13, 2006 1:02 PM> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Dropped Calls on TDM400p These are just PSTN calls, and I have set busydetect=no and> callprogress=no in zapata.conf as per voip-info guidance, but problem
> persists.>> CPU load never breaks 20, so that doesn't seem to be the problem, but> it's a 1.2Ghz Athlon with 768MB RAM.>> Power supply to system is clean, there's no heavy network traffic
> going on besides Asterisk to the phones (Aastra 480i's).>> What other factors can I investigate?>> This client is so unhappy they are ready to go back to their old PBX> system.
>> I am desperate, please help!!>> Thanks in advance,> Jonathan>___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] Building Zaptel 1.2.9 with Octasic

2006-09-13 Thread Steve Kennedy

On Wed, Sep 13, 2006 at 01:34:30PM -0400, Mark Hulber wrote:

> Yes, it worked.  I didn't get the announcement of 1.2.9.1.

Seems it wasn't announced, nor Asterisk 1.2.12.1

Nor their new Asterisk Appliance that seems to run off Flash (with a GUI
that configures it all). ALso the new 4 port BRI card is on the site.


Steve

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Re: [asterisk-users] Dropped Calls on TDM400p


Sorry, see now that it is pots.

How do your interrupts look?  What is the hardware platform or more 
specifically the MB?  Is the platform listed on Digium's site as 
approved or listed as having issues?


Thanks,
Steve

Steven Totaro wrote:


They need to document the exact day and time so you can look in the 
logs.  Is this a T1 or POTS?


 

Customers always complain and threaten to go back to their old PBX.  
First, calm down, then calm them down and make sure they know you are 
working on it.  Every new install is going to have issues that will 
take time to resolve.  Remember that in your pricing or you will soon 
be out of business. 

 

Get exact times and frequency then check your logs to see if anything 
matches that may be an issue. 

 


Thanks,

Steve

 




*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of 
*Jonathan Barratt

*Sent:* Wednesday, September 13, 2006 1:02 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Dropped Calls on TDM400p

 

These are just PSTN calls, and I have set busydetect=no and 
callprogress=no in zapata.conf as per voip-info guidance, but problem 
persists.


CPU load never breaks 20, so that doesn't seem to be the problem, but 
it's a 1.2Ghz Athlon with 768MB RAM.


Power supply to system is clean, there's no heavy network traffic 
going on besides Asterisk to the phones (Aastra 480i's).


What other factors can I investigate? 

This client is so unhappy they are ready to go back to their old PBX 
system.


I am desperate, please help!!

Thanks in advance,
Jonathan



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Re: [asterisk-users] IVR not able to Play the Balance.. need some helphere

I think what matters is what directory the file resides in.  They are 
all wav or gsm.


ram wrote:

Hi
 
is this possible to read words like digits  in asterisk
 
Ram


 
On 9/13/06, *Héctor Maldonado* <[EMAIL PROTECTED] 
> wrote:


uh.. maybe recording a gsm file with "rupees" and playing it just
after SayDigits.. ?
 

 
2006/9/13, ram <[EMAIL PROTECTED] >:


Hi all
 
Same like reading  Numbers

how can read words
 
since i dont see India Currency anouncing ( i see Dollars)

i want to anounce after 17 then Rupees
 
how can i achive this
 
Ram


 
On 9/13/06, *bails* <[EMAIL PROTECTED]

> wrote:

> exten => 888,8,SayDigits(${AMOUNT-DUE})

B

ram wrote:
> Hi
>
> thanks for the quick reply
>
> yes as suggested i did the Following modification
>
> exten => 888,1,Read(${CALLERIDNUM})
> exten => 888,2,MYSQL(Connect connid 127.0.0.1
 < http://127.0.0.1
> root
> password database)
> exten => 888,3,MYSQL(Query resultid ${connid} select\
saldo\ from\
> balance\ where\ username=${CALLERIDNUM})
> exten => 888,4,Wait(1)
> exten => 888,5,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)
> exten => 888,6,Playback(card-balance-is)
> exten => 888,7,Wait(2)
> exten => 888,8,Playback(${AMOUNT-DUE})
> exten => 888,9,MYSQL(Clear ${resultid})
> exten => 888,10,MYSQL(Disconnect ${connid})
>
> when i dial 888, i get sound card balance is,then after
few seconds hangup.
>
> and i see in debug the following error
>
> Sep 13 18:46:59 WARNING[3907]: file.c:512
ast_openstream_full: File 17
> does not exist in any format
> Sep 13 18:46:59 WARNING[3907]: file.c :824
ast_streamfile: Unable to open
> 17 (format gsm): No such file or directory
>
>
> how to play the results
>
> can some one help me in this regard
>
> Ram
>

s
  


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RE: [asterisk-users] Dropped Calls on TDM400p

2006-09-13 Thread Steven Totaro











They need to document the exact day and
time so you can look in the logs.  Is this a T1 or POTS?

 

Customers always complain and threaten to
go back to their old PBX.  First, calm down, then calm them down and make sure
they know you are working on it.  Every new install is going to have issues
that will take time to resolve.  Remember that in your pricing or you will soon
be out of business. 

 

Get exact times and frequency then check
your logs to see if anything matches that may be an issue.  

 

Thanks,

Steve

 













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Barratt
Sent: Wednesday, September 13,
2006 1:02 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped
Calls on TDM400p



 

These are just PSTN calls, and I have set busydetect=no and
callprogress=no in zapata.conf as per voip-info guidance, but problem persists.

CPU load never breaks 20, so that doesn't seem to be the problem, but it's a
1.2Ghz Athlon with 768MB RAM.

Power supply to system is clean, there's no heavy network traffic going on
besides Asterisk to the phones (Aastra 480i's).

What other factors can I investigate?  

This client is so unhappy they are ready to go back to their old PBX system.

I am desperate, please help!!

Thanks in advance,
Jonathan








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Re: [asterisk-users] Building Zaptel 1.2.9 with Octasic

2006-09-13 Thread Mark Hulber


Yes, it worked.  I didn't get the announcement of 1.2.9.1.

MARK.

Tzafrir Cohen wrote:

On Wed, Sep 13, 2006 at 12:00:27PM -0400, Mark Hulber wrote:
  

Any pointers about on how to get around this build problem in Zaptel 1.2.9?



Get 1.2.9.1, that has fixed exactly that.

(and improvd Astribank drivers, thanks Kevin)

  

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Re: [asterisk-users] IVR not able to Play the Balance.. need some helphere

Hi
 
is this possible to read words like digits  in asterisk
 
Ram 
On 9/13/06, Héctor Maldonado <[EMAIL PROTECTED]> wrote:

uh.. maybe recording a gsm file with "rupees" and playing it just after SayDigits.. ?
 
 
2006/9/13, ram <[EMAIL PROTECTED]>: 



Hi all
 
Same like reading  Numbers
how can read words
 
since i dont see India Currency anouncing ( i see Dollars)
i want to anounce after 17 then Rupees
 
how can i achive this

 
Ram 

On 9/13/06, bails <[EMAIL PROTECTED]> wrote: 

> exten => 888,8,SayDigits(${AMOUNT-DUE})Bram wrote:> Hi>> thanks for the quick reply 
>> yes as suggested i did the Following modification>> exten => 888,1,Read(${CALLERIDNUM})> exten => 888,2,MYSQL(Connect connid 
127.0.0.1 < http://127.0.0.1> root> password database)> exten => 888,3,MYSQL(Query resultid ${connid} select\ saldo\ from\ 
> balance\ where\ username=${CALLERIDNUM})> exten => 888,4,Wait(1)> exten => 888,5,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)> exten => 888,6,Playback(card-balance-is)> exten => 888,7,Wait(2) 
> exten => 888,8,Playback(${AMOUNT-DUE})> exten => 888,9,MYSQL(Clear ${resultid}) > exten => 888,10,MYSQL(Disconnect ${connid})>> when i dial 888, i get sound card balance is,then after few seconds hangup. 
>> and i see in debug the following error>> Sep 13 18:46:59 WARNING[3907]: file.c:512 ast_openstream_full: File 17> does not exist in any format> Sep 13 18:46:59 WARNING[3907]: file.c
 :824 ast_streamfile: Unable to open> 17 (format gsm): No such file or directory>> > how to play the results>> can some one help me in this regard>> Ram>> 
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Re: [asterisk-users] Building Zaptel 1.2.9 with Octasic

2006-09-13 Thread Steve Kennedy

On Wed, Sep 13, 2006 at 12:33:01PM -0400, Steven Totaro wrote:

> Use SVN and not the tarball. 

Digium updated to 1.2.9.1 earlier this week.


Steve

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Re: [asterisk-users] IVR not able to Play the Balance.. need some helphere

2006-09-13 Thread Héctor Maldonado

uh.. maybe recording a gsm file with "rupees" and playing it just after SayDigits.. ?
 
 
2006/9/13, ram <[EMAIL PROTECTED]>:


Hi all
 
Same like reading  Numbers
how can read words
 
since i dont see India Currency anouncing ( i see Dollars)
i want to anounce after 17 then Rupees
 
how can i achive this

 
Ram 

On 9/13/06, bails <[EMAIL PROTECTED]> wrote:
 
> exten => 888,8,SayDigits(${AMOUNT-DUE})Bram wrote:> Hi>> thanks for the quick reply 
>> yes as suggested i did the Following modification>> exten => 888,1,Read(${CALLERIDNUM})> exten => 888,2,MYSQL(Connect connid 
127.0.0.1 < http://127.0.0.1> root> password database)> exten => 888,3,MYSQL(Query resultid ${connid} select\ saldo\ from\
> balance\ where\ username=${CALLERIDNUM})> exten => 888,4,Wait(1)> exten => 888,5,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)> exten => 888,6,Playback(card-balance-is)> exten => 888,7,Wait(2)
> exten => 888,8,Playback(${AMOUNT-DUE})> exten => 888,9,MYSQL(Clear ${resultid}) > exten => 888,10,MYSQL(Disconnect ${connid})>> when i dial 888, i get sound card balance is,then after few seconds hangup.
>> and i see in debug the following error>> Sep 13 18:46:59 WARNING[3907]: file.c:512 ast_openstream_full: File 17> does not exist in any format> Sep 13 18:46:59 WARNING[3907]: file.c
:824 ast_streamfile: Unable to open> 17 (format gsm): No such file or directory>> > how to play the results>> can some one help me in this regard>> Ram>>
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[asterisk-users] Dropped Calls on TDM400p

2006-09-13 Thread Jonathan Barratt

These are just PSTN calls, and I have set busydetect=no and
callprogress=no in zapata.conf as per voip-info guidance, but problem
persists.

CPU load never breaks 20, so that doesn't seem to be the problem, but it's a 1.2Ghz Athlon with 768MB RAM.

Power supply to system is clean, there's no heavy network traffic going on besides Asterisk to the phones (Aastra 480i's).

What other factors can I investigate?  

This client is so unhappy they are ready to go back to their old PBX system.

I am desperate, please help!!

Thanks in advance,
Jonathan
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Re: [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?

I tried the had a problem with an NFS dropping while writing audio 
directly from the monitor app and asterisk froze for about 10 seconds 
and then became responsive again.  I write all the recording un-MUXed 
locally and run a cron that ftps them.  We currently have 12 T1s turned 
up, not sure of the simultaneous number of recordings though.


Thanks,
Steve

MF wrote:

Ok I see,   thanks for your info Steve,
Is the Mux problem because of processor time or Disk access time?
because if it is a processor thing,  I should think on a solution 
where,  instead of sending the files over , I just send a command and 
from the other machine Mux them on a mapped drive  (prior to this 
having mapped the directory   where the files are, probably with NFS)   .

Steve Totaro escribió:

Raphaël Jacquot wrote:

MF wrote:
 

Hi,

I have a 2 E1 system with 32 zap FXS extensions (all Zaptel,  with
TDM2400),  on a PIV,  3GHz, 1GB, Well my question is wether I'll be 
able to use it for peak demand moment,

that is having all 60 channels busy 30 talking to agents on the FXS,
while recording their conversation at the same time,  and the other 30
with music on hold while wait.  This is all based on the Queue
application of *.

Does any one thinks this system WONT be able to handle it? Am I
crazy of even trying it?



the intel processors at great at number crunching, but suck ass at I/O

I'd get an amd64 box instead

  
Intel should work fine but do not mux the conversations on the box, i 
put a second NIC in mine on a different network than the VoIP and FTP 
the files to a box that does the MUXing.  We have recorded well over 
60 simultaneous calls.  Use the monitor in the queues app so you do 
not get all the music stuff recorded and you should be just fine.  
Actually, our setups are different though, I have separate boxes that 
just handle the T1 to SIP handoff.  They get 50% CPU utilization with 
95 calls and similar specs to your setup.  I still think you should 
probably be ok though.


Thanks,
Steve Totaro

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[asterisk-users] University switches to Asterisk

2006-09-13 Thread Doug Lytle


Interesting article I found linked from Groklaw:

"Sam Houston State University replaces Cisco CallManagers, Nortel PBXs 
with Linux-based VoIP and messaging servers"


http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1

Doug

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deserve neither Liberty nor Safety."


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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Steve Davies


On 9/13/06, Steve Underwood <[EMAIL PROTECTED]> wrote:

Artifex Maximus wrote:

If you look in http://www.soft-switch.org/download/snapshots/snapdsp,
the latest snapshot of spandsp and the app_rxfax and app_txfax
applications there provide ECM. It is less well tested than the
spandsp-0.0.2 code, but seems to be working pretty well now.



Fantastic.

This looks pretty good I have to say - The ECM seems as if it may be a
little intolerant... On a fax machine where I got 100% success in the
past with 0.0.2, I am now getting "result (60) Disconnected after
permitted retries." on about every 4th page.

Is the ECM tolerance level tuneable in spandsp, or is this
hard-defined in the standard? Is it just a matter of changing:
 #define MAX_MESSAGE_TRIES   3

I also noticed that the page title in the TIFF does not appear to be
set for the last page received any more (I have not looked into this
at-all though, so it may be my environment).

Thanks again.
Steve
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[asterisk-users] callback without agi

2006-09-13 Thread Patricio Valarezo

Hi, it's possible to implement a callback without agi?, i'm trying this 
but * exits without dialing (if I hungup during s,3 wait) but if it 
hungs in s,4 it dials, so is there an explanation to this behavior? 
there is an alternative to do it? just for learning


thanks for your answers

[followme]
exten => s,1,NoOp(Followme me sigue)
exten => s,2,NoOp(El CID es ${CALLERID(num)})
exten => s,3,Wait(4)
exten => s,4,Hangup()

; al cortar debera iniciar la secuencia
exten => h,1,NoOp("${CALLERID(num)} ha cortado")
exten => h,n,NoOp(channel es ${CHANNEL})
exten => h,n,Wait(10)
exten => h,n,NoOp("aqui podriamos marcar")
exten => h,n,Dial(${CANAL}/${CALLERID(num)})
exten => s,n,Hangup()


PV.
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[asterisk-users] Third Lane PBX Manger Multi-Tenant

2006-09-13 Thread Bill Gibbs









Anyone using that product with success?

 

Bill






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RE: [asterisk-users] Building Zaptel 1.2.9 with Octasic

2006-09-13 Thread Steven Totaro

Use SVN and not the tarball. 
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Mark Hulber
> Sent: Wednesday, September 13, 2006 12:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Building Zaptel 1.2.9 with Octasic
> 
> Any pointers about on how to get around this build problem in Zaptel
> 1.2.9?
> 
> /usr/src/zaptel-1.2.9/wct4xxp/fw2h
> /usr/src/zaptel-1.2.9/wct4xxp/OCT6114-128D.ima
> /usr/src/zaptel-1.2.9/wct4xxp/vpm450m_fw.h
> make[3]: *** No rule to make target
> `/usr/src/zaptel-
> 1.2.9/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h',
> needed by `/usr/src/zaptel-1.2.9/wct4xxp/vpm450m.o'.  Stop.
> make[2]: *** [/usr/src/zaptel-1.2.9/wct4xxp] Error 2
> make[1]: *** [_module_/usr/src/zaptel-1.2.9] Error 2
> make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686'
> make: *** [linux26] Error 2
> 
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Re: [asterisk-users] Building Zaptel 1.2.9 with Octasic

2006-09-13 Thread Tzafrir Cohen

On Wed, Sep 13, 2006 at 12:00:27PM -0400, Mark Hulber wrote:
> Any pointers about on how to get around this build problem in Zaptel 1.2.9?

Get 1.2.9.1, that has fixed exactly that.

(and improvd Astribank drivers, thanks Kevin)

-- 
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icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
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Re: [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?

2006-09-13 Thread MF

Ok I see,   thanks for your info Steve, 

Is the Mux problem because of processor time or Disk access time?
because if it is a processor thing,  I should think on a solution 
where,  instead of sending the files over , I just send a command and 
from the other machine Mux them on a mapped drive  (prior to this having 
mapped the directory   where the files are, probably with NFS)   . 


Steve Totaro escribió:

Raphaël Jacquot wrote:

MF wrote:
 

Hi,

I have a 2 E1 system with 32 zap FXS extensions (all Zaptel,  with
TDM2400),  on a PIV,  3GHz, 1GB, Well my question is wether I'll be 
able to use it for peak demand moment,

that is having all 60 channels busy 30 talking to agents on the FXS,
while recording their conversation at the same time,  and the other 30
with music on hold while wait.  This is all based on the Queue
application of *.

Does any one thinks this system WONT be able to handle it? Am I
crazy of even trying it?



the intel processors at great at number crunching, but suck ass at I/O

I'd get an amd64 box instead

  
Intel should work fine but do not mux the conversations on the box, i 
put a second NIC in mine on a different network than the VoIP and FTP 
the files to a box that does the MUXing.  We have recorded well over 
60 simultaneous calls.  Use the monitor in the queues app so you do 
not get all the music stuff recorded and you should be just fine.  
Actually, our setups are different though, I have separate boxes that 
just handle the T1 to SIP handoff.  They get 50% CPU utilization with 
95 calls and similar specs to your setup.  I still think you should 
probably be ok though.


Thanks,
Steve Totaro

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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Steve Underwood


Hi Bruce,

Looks like your typing is as bad as mine :-)

Try http://www.soft-switch.org/downloads/snapshots/spandsp

Steve

Bruce Reeves wrote:

Try, http://www.soft-switch.org/downloads/snapshots/snapdsp 
,


On 9/13/06, *Artifex Maximus* <[EMAIL PROTECTED] 
> wrote:


Hello Steve,

On 9/13/06, Steve Underwood <[EMAIL PROTECTED]
> wrote:
> Artifex Maximus wrote:
>
> > Hello,
> >
> > I had received a lot of unreadable pages with rxfax. I've been
doing
> > some search on net and found this:
> > http://threebit.net/mail-archive/asterisk-users/msg15708.html
> >
> > It looks like rxfax/spandsp doesn't support ecm error
correction. Bad
> > news for me. Is it still the case? app_rxfax.c dated as 8th of
> > february so I think the answer is yes but I am still hoping a
little
> > no or might someone have a patch for enabling/implementing ecm.
>
> If you look in
http://www.soft-switch.org/download/snapshots/snapdsp,
> the latest snapshot of spandsp and the app_rxfax and app_txfax
> applications there provide ECM. It is less well tested than the
> spandsp-0.0.2 code, but seems to be working pretty well now.

Sounds promising but gives me
"Not Found

The requested URL /download/snapshots/snapdsp was not found on
this server.
Apache/2.0.52 (CentOS) Server at www.soft-switch.org
 Port 80"



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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Craig Guy


Try this one:

http://www.soft-switch.org/downloads/snapshots/spandsp/

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


Sent: Wednesday, September 13, 2006 11:33 PM
Subject: Re: [asterisk-users] rxfax, spandsp and lack of ecm



Hello Steve,

On 9/13/06, Steve Underwood <[EMAIL PROTECTED]> wrote:

Artifex Maximus wrote:

> Hello,
>
> I had received a lot of unreadable pages with rxfax. I've been doing
> some search on net and found this:
> http://threebit.net/mail-archive/asterisk-users/msg15708.html
>
> It looks like rxfax/spandsp doesn't support ecm error correction. Bad
> news for me. Is it still the case? app_rxfax.c dated as 8th of
> february so I think the answer is yes but I am still hoping a little
> no or might someone have a patch for enabling/implementing ecm.

If you look in http://www.soft-switch.org/download/snapshots/snapdsp,
the latest snapshot of spandsp and the app_rxfax and app_txfax
applications there provide ECM. It is less well tested than the
spandsp-0.0.2 code, but seems to be working pretty well now.


Sounds promising but gives me
"Not Found

The requested URL /download/snapshots/snapdsp was not found on this 
server.

Apache/2.0.52 (CentOS) Server at www.soft-switch.org Port 80"

bye,
Zsolt
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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Bruce Reeves

Try, http://www.soft-switch.org/downloads/snapshots/snapdsp,
On 9/13/06, Artifex Maximus <[EMAIL PROTECTED]> wrote:
Hello Steve,On 9/13/06, Steve Underwood <[EMAIL PROTECTED]> wrote:> Artifex Maximus wrote:>> > Hello,> >> > I had received a lot of unreadable pages with rxfax. I've been doing
> > some search on net and found this:> > http://threebit.net/mail-archive/asterisk-users/msg15708.html> >> > It looks like rxfax/spandsp doesn't support ecm error correction. Bad
> > news for me. Is it still the case? app_rxfax.c dated as 8th of> > february so I think the answer is yes but I am still hoping a little> > no or might someone have a patch for enabling/implementing ecm.
>> If you look in http://www.soft-switch.org/download/snapshots/snapdsp,> the latest snapshot of spandsp and the app_rxfax and app_txfax
> applications there provide ECM. It is less well tested than the> spandsp-0.0.2 code, but seems to be working pretty well now.Sounds promising but gives me"Not FoundThe requested URL /download/snapshots/snapdsp was not found on this server.
Apache/2.0.52 (CentOS) Server at www.soft-switch.org Port 80"bye,Zsolt___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya

BJ Weschke wrote:

On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

BJ Weschke wrote:
> On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> Doug Lytle wrote:
>> > Steve Totaro wrote:
>> >> I am trying to connect a Definity G3 to an asterisk system.  I 
had it

>> >> working OK with the exception of the caller ID on the Definity
>> >> handsets just
>> >
>> > He wants to know if your Definity is an S, SI or an R?
>> >
>> > Doug
>> >
>> I am not sure, it is a refrigerator sized unit with three 
cabinets.  The
>> manual was printed in 1988 and is ATT&T Definity Communications 
System

>> Generic1 and Generic3.
>>
>> If you need more, I can go to the remote system and provide more
>> details.
>>
>> I REALLY appreciate you helping on this.
>>
>
> Sounds like a G3R. How are you signaling between the two? PRI?
>
Yes.  I have two ISDN communication processor links setup which I later
read on google was not supported.  Now I cannot remove the new entry.  I
just get "identifier not assigned"

What you'll want to do in order to set this up, is build a trunk
group in DSA that consists of the two T1's, and then you'll want to
build a signaling group that uses "isdn" signaling and then (probably
3-4 pages over in DSA) you'll want to assign all the channels of those
T1s as "members" of that signaling group.

On the asterisk side you'll want to setup the trunks as pri_net.

Put a T1 crossover cable in between, if there's no carrier loops
involved, and then you're done.

I had one working ok, then I added a second.  The problem (I think) is 
that I have two communication processor links set to ISDN and through 
google I have found that you can only have one.  I cannot remove the second.

Are you saying that it is ok to have two ISDN communication processor 
links if I setup the way you are describing?  What is DSA?  It sounds 
like you are describing an NFAS setup.

Thanks
Steve
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[asterisk-users] Building Zaptel 1.2.9 with Octasic

2006-09-13 Thread Mark Hulber


Any pointers about on how to get around this build problem in Zaptel 1.2.9?

   /usr/src/zaptel-1.2.9/wct4xxp/fw2h
   /usr/src/zaptel-1.2.9/wct4xxp/OCT6114-128D.ima
   /usr/src/zaptel-1.2.9/wct4xxp/vpm450m_fw.h
   make[3]: *** No rule to make target
   `/usr/src/zaptel-1.2.9/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h',
   needed by `/usr/src/zaptel-1.2.9/wct4xxp/vpm450m.o'.  Stop.
   make[2]: *** [/usr/src/zaptel-1.2.9/wct4xxp] Error 2
   make[1]: *** [_module_/usr/src/zaptel-1.2.9] Error 2
   make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686'
   make: *** [linux26] Error 2

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Re: [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?


Raphaël Jacquot wrote:

MF wrote:
  

Hi,

I have a 2 E1 system with 32 zap FXS extensions (all Zaptel,  with
TDM2400),  on a PIV,  3GHz, 1GB, 
Well my question is wether I'll be able to use it for peak demand moment,

that is having all 60 channels busy 30 talking to agents on the FXS,
while recording their conversation at the same time,  and the other 30
with music on hold while wait.  This is all based on the Queue
application of *.

Does any one thinks this system WONT be able to handle it? Am I
crazy of even trying it?



the intel processors at great at number crunching, but suck ass at I/O

I'd get an amd64 box instead

  
Intel should work fine but do not mux the conversations on the box, i 
put a second NIC in mine on a different network than the VoIP and FTP 
the files to a box that does the MUXing.  We have recorded well over 60 
simultaneous calls.  Use the monitor in the queues app so you do not get 
all the music stuff recorded and you should be just fine.  Actually, our 
setups are different though, I have separate boxes that just handle the 
T1 to SIP handoff.  They get 50% CPU utilization with 95 calls and 
similar specs to your setup.  I still think you should probably be ok 
though.


Thanks,
Steve Totaro

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Re: [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?

2006-09-13 Thread MF


Thanks Raphael

I'll plan to upgrade to amd64 right away, but in the mean time do you 
think working with PIV could be a major problem?   that is is it 
probable that this setting will miserably fail with a PIV ??  Or 
just a tight situation where it might work  with some limitations when 
fully loaded?   
What should I expect while working with PIV before switching to amd64?.


thanks

Manrique

Raphaël Jacquot escribió:

MF wrote:
  

Hi,

I have a 2 E1 system with 32 zap FXS extensions (all Zaptel,  with
TDM2400),  on a PIV,  3GHz, 1GB, 
Well my question is wether I'll be able to use it for peak demand moment,

that is having all 60 channels busy 30 talking to agents on the FXS,
while recording their conversation at the same time,  and the other 30
with music on hold while wait.  This is all based on the Queue
application of *.

Does any one thinks this system WONT be able to handle it? Am I
crazy of even trying it?



the intel processors at great at number crunching, but suck ass at I/O

I'd get an amd64 box instead

  

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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Artifex Maximus

Hello Steve,

On 9/13/06, Steve Underwood <[EMAIL PROTECTED]> wrote:

Artifex Maximus wrote:

> Hello,
>
> I had received a lot of unreadable pages with rxfax. I've been doing
> some search on net and found this:
> http://threebit.net/mail-archive/asterisk-users/msg15708.html
>
> It looks like rxfax/spandsp doesn't support ecm error correction. Bad
> news for me. Is it still the case? app_rxfax.c dated as 8th of
> february so I think the answer is yes but I am still hoping a little
> no or might someone have a patch for enabling/implementing ecm.

If you look in http://www.soft-switch.org/download/snapshots/snapdsp,
the latest snapshot of spandsp and the app_rxfax and app_txfax
applications there provide ECM. It is less well tested than the
spandsp-0.0.2 code, but seems to be working pretty well now.

Sounds promising but gives me
"Not Found

The requested URL /download/snapshots/snapdsp was not found on this server.
Apache/2.0.52 (CentOS) Server at www.soft-switch.org Port 80"

bye,
Zsolt
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[asterisk-users] set global variable

2006-09-13 Thread Jan Fousek

Hi all,
 is there any possibility of setting the global variables from outside of 
asterisk?
Like manager api or something like that.

Thanks a lot

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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Steve Underwood


Artifex Maximus wrote:


Hello,

I had received a lot of unreadable pages with rxfax. I've been doing
some search on net and found this:
http://threebit.net/mail-archive/asterisk-users/msg15708.html

It looks like rxfax/spandsp doesn't support ecm error correction. Bad
news for me. Is it still the case? app_rxfax.c dated as 8th of
february so I think the answer is yes but I am still hoping a little
no or might someone have a patch for enabling/implementing ecm.


If you look in http://www.soft-switch.org/download/snapshots/snapdsp, 
the latest snapshot of spandsp and the app_rxfax and app_txfax 
applications there provide ECM. It is less well tested than the 
spandsp-0.0.2 code, but seems to be working pretty well now.


Steve

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[asterisk-users] audio drop out half channel

2006-09-13 Thread Jerry Geis

I am having half channel audio after about 6 minutes. asterisk 1.2.11 
and 1.2.8 zaptel.

My call is originating on a a SIP extension going out to nufone to my cell.
After about 6 minutes the SIP extension can still hear me on my cell
but I cannot hear the SIP extension.

If the SIP extension hits a key I hear it on my cell.
If the SIP extension is talking and hits hold the cell hears about 2 
seconds of audio then

nothing again.

What might this be or what should I do to find out what is going on.

Jerry
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RE: [asterisk-users] voicemailmain errors on CLI

2006-09-13 Thread Sergio R. D'Ippolito

You have to leave a message in the voicemail, then listen it and the error
will not apear again.

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Doug Lytle
Enviado el: Miércoles, 13 de Septiembre de 2006 08:45 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] voicemailmain errors on CLI

Benjamin Jacob wrote:
> Hello ppl,
> I am getting the following errors when accessing voicemails
> Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to 
> create lock file 
> '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file 
> or directory

Just as the error states, the directory  Old doesn't exist.  Check to 
see if it does.  If it is there, check it's permissions, if not then 
create it.

Doug


-- 
 
Ben Franklin quote:

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


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Re: [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?

2006-09-13 Thread Raphaël Jacquot

MF wrote:
> Hi,
> 
> I have a 2 E1 system with 32 zap FXS extensions (all Zaptel,  with
> TDM2400),  on a PIV,  3GHz, 1GB, 
> Well my question is wether I'll be able to use it for peak demand moment,
> that is having all 60 channels busy 30 talking to agents on the FXS,
> while recording their conversation at the same time,  and the other 30
> with music on hold while wait.  This is all based on the Queue
> application of *.
> 
> Does any one thinks this system WONT be able to handle it? Am I
> crazy of even trying it?

the intel processors at great at number crunching, but suck ass at I/O

I'd get an amd64 box instead
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Re: [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya

2006-09-13 Thread BJ Weschke

On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

BJ Weschke wrote:
> On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> Doug Lytle wrote:
>> > Steve Totaro wrote:
>> >> I am trying to connect a Definity G3 to an asterisk system.  I had it
>> >> working OK with the exception of the caller ID on the Definity
>> >> handsets just
>> >
>> > He wants to know if your Definity is an S, SI or an R?
>> >
>> > Doug
>> >
>> I am not sure, it is a refrigerator sized unit with three cabinets.  The
>> manual was printed in 1988 and is ATT&T Definity Communications System
>> Generic1 and Generic3.
>>
>> If you need more, I can go to the remote system and provide more
>> details.
>>
>> I REALLY appreciate you helping on this.
>>
>
> Sounds like a G3R. How are you signaling between the two? PRI?
>
Yes.  I have two ISDN communication processor links setup which I later
read on google was not supported.  Now I cannot remove the new entry.  I
just get "identifier not assigned"

What you'll want to do in order to set this up, is build a trunk
group in DSA that consists of the two T1's, and then you'll want to
build a signaling group that uses "isdn" signaling and then (probably
3-4 pages over in DSA) you'll want to assign all the channels of those
T1s as "members" of that signaling group.

On the asterisk side you'll want to setup the trunks as pri_net.

Put a T1 crossover cable in between, if there's no carrier loops
involved, and then you're done.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] ip address incoming call

2006-09-13 Thread antonio




Hi, is there a 
variable hold the ip address of the incoming sip call ?
Thanks
 
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Re: [asterisk-users] rxfax, spandsp and lack of ecm

2006-09-13 Thread Craig Guy

It would be nice if someone could do that but I doubt it will happen. 
Hylafax / iaxmodem is more complicated and more effort to set up than rxfax 
but the end result is worth the effort.  My only criticism is that I set up 
2 x E1's on a server (60 channels) and I didn't enjoy having to configure 60 
entries in iax.conf, 60 tty's in etc/inittab, 60 modem entries in 
var/spool/hylafax/etc, 60 entries in extensions.conf .. you get the picture. 
In that respect rxfax is much much easier and faster to get going, and also 
more scalable cause you can just keep calling it as many times as you need 
it without having to know your max concurrent calls in advance.

Craig

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

Sent: Wednesday, September 13, 2006 8:40 PM
Subject: Re: [asterisk-users] rxfax, spandsp and lack of ecm

Craig & Doug,

Thanks for your info. I'll do that way.

Is there any chance for implementing ecm in rcfax/spandsp? I think
using rxfax is more friendly than using a modem emulator connected
through a virtual device to a fax software. It's sound as a very
bizarre way to me. :-)

bye,
Zsolt

On 9/13/06, Craig Guy <[EMAIL PROTECTED]> wrote:

spandsp supports 9600 rx and does not support ecm.  If you want ecm, use
iaxmodem with hylafax - http://iaxmodem.sourceforge.net , currently 
hylafax

in conjunction with iaxmodem seems to be more reliable than rxfax and
spandsp by themselves.

Craig
- Original Message -
From: "Artifex Maximus" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, September 13, 2006 6:31 PM
Subject: [asterisk-users] rxfax, spandsp and lack of ecm

> Hello,
>
> I had received a lot of unreadable pages with rxfax. I've been doing
> some search on net and found this:
> http://threebit.net/mail-archive/asterisk-users/msg15708.html
>
> It looks like rxfax/spandsp doesn't support ecm error correction. Bad
> news for me. Is it still the case? app_rxfax.c dated as 8th of
> february so I think the answer is yes but I am still hoping a little
> no or might someone have a patch for enabling/implementing ecm.
>
> bye,
> Zsolt

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Re: [asterisk-users] IVR not able to Play the Balance.. need some helphere

Hi all
 
Same like reading  Numbers
how can read words
 
since i dont see India Currency anouncing ( i see Dollars)
i want to anounce after 17 then Rupees
 
how can i achive this
 
Ram 
On 9/13/06, bails <[EMAIL PROTECTED]> wrote:
> exten => 888,8,SayDigits(${AMOUNT-DUE})Bram wrote:> Hi>> thanks for the quick reply
>> yes as suggested i did the Following modification>> exten => 888,1,Read(${CALLERIDNUM})> exten => 888,2,MYSQL(Connect connid 127.0.0.1 <
http://127.0.0.1> root> password database)> exten => 888,3,MYSQL(Query resultid ${connid} select\ saldo\ from\> balance\ where\ username=${CALLERIDNUM})> exten => 888,4,Wait(1)
> exten => 888,5,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)> exten => 888,6,Playback(card-balance-is)> exten => 888,7,Wait(2)> exten => 888,8,Playback(${AMOUNT-DUE})> exten => 888,9,MYSQL(Clear ${resultid})
> exten => 888,10,MYSQL(Disconnect ${connid})>> when i dial 888, i get sound card balance is,then after few seconds hangup.>> and i see in debug the following error>> Sep 13 18:46:59 WARNING[3907]: 
file.c:512 ast_openstream_full: File 17> does not exist in any format> Sep 13 18:46:59 WARNING[3907]: file.c:824 ast_streamfile: Unable to open> 17 (format gsm): No such file or directory>>
> how to play the results>> can some one help me in this regard>> Ram>>> >> ___
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[asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?

2006-09-13 Thread MF


Hi,

I have a 2 E1 system with 32 zap FXS extensions (all Zaptel,  with 
TDM2400),  on a PIV,  3GHz, 1GB,  

Well my question is wether I'll be able to use it for peak demand moment, 

that is having all 60 channels busy 30 talking to agents on the FXS, 
while recording their conversation at the same time,  and the other 30 
with music on hold while wait.  This is all based on the Queue 
application of *.


Does any one thinks this system WONT be able to handle it? Am I 
crazy of even trying it?

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Re: [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya

BJ Weschke wrote:

On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

Doug Lytle wrote:
> Steve Totaro wrote:
>> I am trying to connect a Definity G3 to an asterisk system.  I had it
>> working OK with the exception of the caller ID on the Definity
>> handsets just
>
> He wants to know if your Definity is an S, SI or an R?
>
> Doug
>
I am not sure, it is a refrigerator sized unit with three cabinets.  The
manual was printed in 1988 and is ATT&T Definity Communications System
Generic1 and Generic3.

If you need more, I can go to the remote system and provide more 
details.

I REALLY appreciate you helping on this.

Sounds like a G3R. How are you signaling between the two? PRI?

Yes.  I have two ISDN communication processor links setup which I later 
read on google was not supported.  Now I cannot remove the new entry.  I 
just get "identifier not assigned"

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Re: [asterisk-users] IVR not able to Play the Balance.. need some helphere


ram wrote:

Hi
 
thanks for the quick reply
 
yes as suggested i did the Following modification
 
exten => 888,1,Read(${CALLERIDNUM})
exten => 888,2,MYSQL(Connect connid 127.0.0.1  root 
password database)
exten => 888,3,MYSQL(Query resultid ${connid} select\ saldo\ from\ 
balance\ where\ username=${CALLERIDNUM})

exten => 888,4,Wait(1)
exten => 888,5,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)
exten => 888,6,Playback(card-balance-is)
exten => 888,7,Wait(2)
exten => 888,8,Playback(${AMOUNT-DUE})
exten => 888,9,MYSQL(Clear ${resultid})
exten => 888,10,MYSQL(Disconnect ${connid})
 
when i dial 888, i get sound card balance is,then after few seconds 
hangup.
 
and i see in debug the following error
 
Sep 13 18:46:59 WARNING[3907]: file.c:512 ast_openstream_full: File 17 
does not exist in any format
Sep 13 18:46:59 WARNING[3907]: file.c:824 ast_streamfile: Unable to 
open 17 (format gsm): No such file or directory


 
how to play the results
 
can some one help me in this regard
 
Ram
Maybe if you copy 17.gsm from /sounds/digits to /sounds it might work or 
put the path in the playback priority 8.


Thanks,
Steve Totaro

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Re: [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya

2006-09-13 Thread Doug Lytle


BJ Weschke wrote:

On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

Sounds like a G3R. How are you signaling between the two? PRI?


Actually, we've discovered it's a DefinityG3 V4

Doug


--

Ben Franklin quote:

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


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Re: [asterisk-users] IVR not able to Play the Balance.. need some

Hi
 
thanks i have rectified the problem
 
ram 
On 9/13/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
ram wrote:>> exten => 888,8,Playback(${AMOUNT-DUE})Change this from Playback to SayNumber
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SayNumberDoug--Ben Franklin quote:"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
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Re: [asterisk-users] IVR not able to Play the Balance.. need some helphere