Re: [Asterisk-Users] Large country based dialplan

[Dinesh Nair]

On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following:

Where I got the data from and all is also on that page if anyone wanted
to make their own lists.  I would appreciate any updates or corrections
that anyone happens to notice.  


a simple modification which would make this a lot more international 
friendly would be the definition of a variable to hold the international 
access code and then using this code instead of _011 which is US-centric.


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Re: [Asterisk-Users] Voicemail Passwords and RealTime

[Tzafrir Cohen]

On Tue, Oct 11, 2005 at 05:37:12PM -0600, Ryan Hulsker wrote:

 
 mine looks like this
 
 #!/usr/bin/perl
 # Takes 3 command line args, context, mailbox, password
 # updates the mailbox password in mysql
 
 use strict;
 use DBI;
 
 my ($Context, $MailBox, $Password) = @ARGV;
 
 my $dbh =
 DBI-connect(dbi:mysql:hostname=localhost;database=asterisk,username, 
 password);
 
 $dbh-do(update voicemail_users set password = '$Password' where
 context = '$Context' and mailbox = '$MailBox');
 
 $dbh-disconnect();

OT:

Why do people resort to perl just for such a simple script?

#!/bin/sh
mysql asterisk -e update voicemail_users set password = '$3' where context = 
'$1' and MailBox = '$2'

Password/hostname, etc. can wither be hard-wired at the command-line or 
using my.cnf (or an alternative my.cnf in the command-line parameters).

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[EMAIL PROTECTED] |   |  best
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Re: [Asterisk-Users] help with broken voicemail

[Tzafrir Cohen]

On Tue, Oct 11, 2005 at 10:09:34PM -0400, Andy Goss wrote:
 Update - 
 
 I made a backup of my entire voicemail directory then deleted it.  If I
 then try and record a greeting, it works.  Asterisk creates the folder
 structure and records the greeting.  If I try to copy the old file back
 into the directory, it wont work.  It's the same file name and
 everything.  The only thing I can figure might be an issue is that the
 voicemail drive is mounted as msdos so maybe there is something
 permissions different about the files that I cant see.  
 
 Any help would be appreciated.  

Please post the output of the following two commands:

ls -l /path/to/message.wav
file  /path/to/message.wav

Is it indeed a valid wav/RIFF file?

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Re: [Asterisk-Users] error message when accessing voicemail

[Tzafrir Cohen]

On Wed, Oct 12, 2005 at 01:44:34AM -0400, Andy Goss wrote:
 If anyone could tell me what this error is all about, I would be very
 grateful.  
 
 Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
 path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not
 permitted
 Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
 path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not
 permitted
 
 Now, goodnight and thank you in advance

Under what user does Asterisk run?

ls -la /var/spool/asterisk/voicemail/default/5933

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Re: [Asterisk-Users] Large country based dialplan

[Erik]

Dinesh Nair wrote:
 
 On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following:
 
 Where I got the data from and all is also on that page if anyone wanted
 to make their own lists.  I would appreciate any updates or corrections
 that anyone happens to notice.  
 
 
 a simple modification which would make this a lot more international
 friendly would be the definition of a variable to hold the international
 access code and then using this code instead of _011 which is US-centric.
 

Seems to be missing a lot of extensions for the Netherlands and my own region 
code is listed as KPN Mobile :)
There is however a complete list of region codes for the netherlands (this one 
is old, but should give you an idea)
http://www.ez.nl/content.jsp?objectid=26840



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Re: [Asterisk-Users] Areski Calling Card GUI

[Garth Summey]

If you haven't seen it already, this will be a lot of help to you.

http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Application+The+idiots+guideV2

You should now be on step 12. :)


G


Omar McKenzie wrote:

Hi

I have gone thru the steps of installing AreskiCC, I would 
like to know how to get access to the GUI interface of this application.





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Re: [Asterisk-Users] Dial DTMF after bridging call

[Corey Frang]

I don't think that its the D() dialing before the call is bridged, I 
just tested it on Asterisk 1.0.7 and CVS HEAD


Both times I did:

Dial(SIP/[EMAIL PROTECTED],20,D(ww1234))

both times i picked up the phone, it waited about 1 second, dialed 1, 
then stopped alltogether.


This might be an actual bug...

Interestingly on CVS HEAD i tried

Dial(SIP/[EMAIL PROTECTED],20,D(ww1234:ww1234))

I heard the DTMF's on the calling phone... I'm wondering if there is 
some issue with how its writing the DTMF to the outgoing SIP channels?


The lucentbox is a MaxTNT.

Interestingly, I started playing with the numbers on my phone after the 
dial messed up, and I could get the DTMF tones stuck playing one tone 
for a long time.  If i took the D() out of it It didn't have that 
problem.


On Aug 25, 2005, at 15:04, Joseph wrote:


Is there a way to dial DTMF after bridging the call.
The current option D() in Dial will dial DTMF before the call is 
bridged

and this doesn't do the job.
I need to dial DTMF after the call is bridged and the message is played
with Background

--
#Joseph
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[Asterisk-Users] unloading TE110P bristuffed module cause kernel panic

[Francesco Angi]

Hi folks,

I've already searched the mailing list but no one else
seems to have my same problem.

I'm using Asterisk with the following configuration:



Fedora Core 4 (but I also tried Fedora 3)



1 Digium TE110P

1 TDM40B

1 HFC-S 'Cologne'



bristuff 0.2.0-RC8o (zaptel 1.0.9.2)



I compiled right, I can load kernel modules but when I
try to unload wcte11xp module (the one for TE110P card) I get a kernel panic: 

Kernel panic - not syncing:
/usr/src/bristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c:333:
spin_lock(usr/src/zbristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c:d9640004)
already locked by /usr/src/bristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c/887.
(Not tainted)



This happens if I load and unload by zaptel script or
if modprobe or insmod 'by hand', then run ztcfg and the unload the module.

No bristuffed zaptel works right and bristuffed zaptel
module for TDM40B works right.



The card does not share IRQ with other devices,
anyway I tried to have only TE110P mounted on PCI slot and to change PCI slot
where card is mounted. Nothing to do.



I really don't know what else I can try.



Thanks for help,

_fangi_






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Re: [Asterisk-Users] unloading TE110P bristuffed module cause kernel panic

[Simone Cittadini]

Francesco Angi ha scritto:


Hi folks,

I've already searched the mailing list but no one else seems to have 
my same problem.


I'm using Asterisk with the following configuration:

 


Fedora Core 4 (but I also tried Fedora 3)

 


1 Digium TE110P

1 TDM40B

1 HFC-S 'Cologne'

 


bristuff 0.2.0-RC8o (zaptel 1.0.9.2)

 

I compiled right, I can load kernel modules but when I try to unload 
wcte11xp module (the one for TE110P card) I get a kernel panic:


Kernel panic - not syncing: 
/usr/src/bristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c:333: 
spin_lock(usr/src/zbristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c:d9640004) 
already locked by 
/usr/src/bristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c/887. (Not tainted)


 

This happens if I load and unload by zaptel script or if modprobe or 
insmod 'by hand', then run ztcfg and the unload the module.


No bristuffed zaptel works right and bristuffed zaptel module for 
TDM40B works right.


 

The card does not share IRQ with other devices, anyway  I tried to 
have only TE110P mounted on PCI slot and to change PCI slot where card 
is mounted. Nothing to do.


 


I really don't know what else I can try.

 


Thanks for help,

_fangi_

Same problem with debian sarge on a dell and asterisk 1.0.7 from 
packages, unloading the module freezes the system, (rebooting the 
machine worked right), I installed zaptel 1.2beta and it seems to work, 
but I haven't really tested it, only loaded/unloaded/loaded and placed a 
couple of calls.

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[Asterisk-Users] unsubscribe

[Sean Rima]

Hello asterisk-users,

  

Sean
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+---+
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RE: [Asterisk-Users] SS7 with Asterisk

[Goran Skular]

Name of the company is MULTI-line GmbH

You can contact mr. Zlatko Medibach at +43 1 78932320 or on cellular +43 676
3220262.
Mail: [EMAIL PROTECTED]

Their HQ is in Wien..

I can not help you with the details, I just know that they implemented SS7
on * for some telcos there.

Goran

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Johann Steinwendtner
Sent: Tuesday, October 11, 2005 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SS7 with Asterisk

Goran,

which company ist this ? Do they use the www.ss7box.com
approach ?

Thanks and best regards

Hans



Goran Skular schrieb:

anyone running SS7 with Asterisk ? Please help me out.
I need to know the hardware used for SS7 with Digium E1 cards...



I can point you to one company in Austria. They deployed SS7 on Asterisk,
but not with Digium cards for one smaller telco.




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[Asterisk-Users] Modifying cmd VoicemailMain

[Kuniyoshi Murata]

Dear Asterisk Users,

I'm a Japanese and now configuring Voicemail.
Now I need to modify the way cmd VoicemailMain works to fix language
difference and other my conveniences.

What I want to do are...

1) Add words used in message retrieving guidance.
I need to add different suffixes to numeric words due to Japanese way of
mentioning time. (e.g. in English, you can say Five forty-five for 5:45,
but in Japanese, we have to put hour and minute for respective time unit
(meaning, VoicemailMain should pronounce as Five hours and forty-five
minutes in Japanese). So, is there any way to add words modifying the
regular word order?

2) Disable most of the key function guidance for retrieving the message.
I don't want too much function guidance of VoicemailMain saying such as 3
for advanced options and the like. I just want to hear just a few important
keys to press. So, is there any way I can separately disable guidance for
each key functions

Any input is welcome.

TIA
Kuni

--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Macintosh Webcast Specialisthttp://www.macwebcaster.com



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Re: [Asterisk-Users] Large country based dialplan

[Piotr Chytla]

On Wed, Oct 12, 2005 at 09:13:44AM +0200, Erik wrote:
 Dinesh Nair wrote:
  
  On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following:
  
  Where I got the data from and all is also on that page if anyone wanted
  to make their own lists.  I would appreciate any updates or corrections
  that anyone happens to notice.  
  
  
  a simple modification which would make this a lot more international
  friendly would be the definition of a variable to hold the international
  access code and then using this code instead of _011 which is US-centric.
  
 
 Seems to be missing a lot of extensions for the Netherlands and my own region 
 code is listed as KPN Mobile :)

The same for Poland, in list I've found only 6 major cities in Poland
(Krakow/Rzeszow/Warsaw/Katowice/Gdansk/Wroclaw) but there is lot 
more zones :

http://www.itu.int/itudoc/itu-t/number/p/pol/81563_ww9.doc

or this :

http://www.ertel.com.pl/python/prefkraj.py

/pch

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[Asterisk-Users] Outgoing Provider Recommendations

[Dan Journo]

Hi,

Can anyone recommend a cheap but reliable company to teminate my asterisk sip calls in Israel (mobile/cell)?

If its against the rules to discuss this on the list, please email it me directly.

Thanks
Dan
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[Asterisk-Users] Voicemail recording volume control

[Kuniyoshi Murata]

Dear Asterisk users,

Other than VoicemailMain, which Im asking in the other mail, I have another
thing to fix.

That is low recording volume of Voicemail. Compared with sound files, volume
in other phone devices that pick up the same kind of phonecall, obviously
the sound level of sound file recorded of voicemail system is low.

Is there any parameters to fix the recording level of voicemail?

TIA
Kuni 

--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Macintosh Webcast Specialisthttp://www.macwebcaster.com



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[Asterisk-Users] delays with IAX2 and Meetme

[Steven Langley]

Hi there



I am using IAX2 softphones dialing into meetme conferences.
I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I
am having is that as soon as there is a delay from a participant, then the delay
continues until the participant hangs up and dials in again. When dialing in
again the delay seems to go.



It seems to me as though as soon as the server registers a
delay from a participant, then it causes delay on all further packets from that
participant.



Does anyone have any ideas what the problem could be?



Many thanks



Steven






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[Asterisk-Users] detect SIP phone availability before dialing

[[EMAIL PROTECTED]]

Hello,

 I need to detect availability of SIP phone before dialing. I need to
know if phone is 
 BUSY, CHANUNAVAIL before dialing. If phone is free, then I will dial
it.

 I need for automatic callback (.call files), but I need to know if it
is available both
 SIP phones before calling.

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[Asterisk-Users] MWI for endpoints not registered at Asterisk

[Stojan Sljivic - Pamet]

Title: Message



Hi,

We 
have phones registered at another soft switch, and would like to use Asterisk as 
a Voicemail system.
Is it 
possible and how to configure Asterisk to send NOTIFY messages (for MWI) to the 
endpoints that are not registered to the Asterisk?

Regards,
Stojan 
Sljivic
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[Asterisk-Users] asterisk log

[asterisk]

Is there a way to

1) disable asterisk from writing in the full  log  ? (
/var/log/asterisk/full )
  or
2) implement a log rotation or similar of the full log ?

I see the full log grows a lot (about 100 MB per Month)

thanks in advance,

Andrea

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[Asterisk-Users] E400P vs te410p vs te411p

[Robert Rozman]

Hi,

I found E400P quad PRI card quite cheap (749USD):

http://www.govarion.com/product_info.php?cPath=1products_id=2osCsid=68cdd6e3d08754

in comparison to te410p (approx 1500 USD )

http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P

Now newer generation with HW echo canceling emerged (te411p).

I'm not sure in what things those two cards differ and what would be best 
option to buy (I believe there is big performance gap between them, but 
don't know how big and if it's worth of money) Also how do you find HW 
echo canceling in te411p ?


Any advice, help ?

Thanks in advance,

regards,

Rob.

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[Asterisk-Users] Asterisk At Home Snom Hints

[Armin Lediger]

Hi, everybody.

I don´t know if it is an * or an AAH issue - I can´t get the
Snom-Phone-hints working under AAH 1.5 running * 1.0.9. I tried with the
Snom 360 softphone and it just doesn´t work.

Is there any known issue?

Is there a AAH mailing list available?

Thank you in advance.

Best regards,
Armin Lediger

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Re: [Asterisk-Users] asterisk log

[gincantalupo]

Hi,

[EMAIL PROTECTED] wrote:


Is there a way to

1) disable asterisk from writing in the full  log  ? (
/var/log/asterisk/full )
 


Take a look at /etc/asterisk/logger.conf


 or
2) implement a log rotation or similar of the full log ?

I see the full log grows a lot (about 100 MB per Month)
 


Use logrotate.


thanks in advance,

Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

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Re: [Asterisk-Users] asterisk log

[Matt Riddell]

[EMAIL PROTECTED] wrote:
 Is there a way to
 
 1) disable asterisk from writing in the full  log  ? (
 /var/log/asterisk/full )

Have a look at /etc/asterisk/logger.conf

 2) implement a log rotation or similar of the full log ?
 
 I see the full log grows a lot (about 100 MB per Month)

Have a look at the CLI command logger rotate and the asterisk -rx command

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] unloading TE110P bristuffed module cause kernel panic

[Tzafrir Cohen]

On Wed, Oct 12, 2005 at 10:18:15AM +0200, Simone Cittadini wrote:

 Same problem with debian sarge on a dell and asterisk 1.0.7 from 
 packages, unloading the module freezes the system, (rebooting the 
 machine worked right), I installed zaptel 1.2beta and it seems to work, 
 but I haven't really tested it, only loaded/unloaded/loaded and placed a 
 couple of calls.

Interesting. The zaptel part of the bristuff patch is rather small and
does not seem to have much to do with locks or with the init code, at
first glance.

Also note that the zaptel patch actually applies cleanly to 1.2 .
Contact me by email for packages. Though I suspect that it is not going
to solve the problem.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Re: delays with IAX2 and Meetme

[Tony Mountifield]

In article [EMAIL PROTECTED],
Steven Langley [EMAIL PROTECTED] wrote:
 
 I am using IAX2 softphones dialing into meetme conferences. I also have
 jitterbuffer=yes, with typical jitterbuffer settings. The problem I am
 having is that as soon as there is a delay from a participant, then the
 delay continues until the participant hangs up and dials in again. When
 dialing in again the delay seems to go.
 
 It seems to me as though as soon as the server registers a delay from a
 participant, then it causes delay on all further packets from that
 participant.
 
 Does anyone have any ideas what the problem could be?

Yes, there are a few possibilities. Firstly, are you using ztdummy for
timing? Which kernel version? If 2.6, have you ensured that USE_RTC is
correctly defined in ztdummy.c?

Look in bugs.digium.com at bug IDs 3599 and 4252 - they might be relevant.

Yesterday I found another mechanism which could give rise to both a delay
and broken audio - I found it with OH323 channels, but it might possibly
arise on other channel types too. It concerns a backlog building up in
the channel driver and never being drained by meetme because of blocking
in the pseudo-device when trying to write the contents of a large frame.

In app_meetme.c, try replacing this:

 careful_write(fd, f-data, f-datalen);

with this:

 if (f-datalen = CONF_SIZE) {
   careful_write(fd, f-data, f-datalen);
 } else {
   ast_log(LOG_WARNING, Discarding large frame (%d bytes) from channel %s\n, 
f-datalen, chan-name);
 }

and see if it helps.

I haven't yet submitted the above change to mantis.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] tx(rx)_fax for *-1.2.0.beta

[Bohuslav Coufal]

Sorry, I could not find it there. I found only version for *-1.1.0.
Could You send right URL to me.

Thanks,

Bob.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roman
Sent: Friday, October 07, 2005 4:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] tx(rx)_fax for *-1.2.0.beta

On Friday 07 October 2005 13:52, Bohuslav Coufal wrote:
 Hi all,

 does anybody have $subj apps.

 Thanks,

 Bob.

you can download them from spandsp website
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[Asterisk-Users] Multiple IAX listeners?

[Leigh Fereday]

We're using Asterisk 
with IAX soft phones to provide communication back to our central office when 
our people travel.

We have configured our 
firewall to allow UDP 4569 forwarded to Asterisk, and have tested this, works no 
problem.

My question is, will 
this support more than 1 simultaneous connection from the same outside IP 
address, or will only one soft phone 
function?

or, put another 
way:

Can multiple soft 
phones (running on separate computers) be used simultaneously from the same 
outside IP address?

TIALeigh
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Re: [Asterisk-Users] supermicro with asterisk and tdm cards

[Patrick]

On Wed, 2005-10-12 at 00:59 -0400, Cory Andrews wrote:
[snip]
 * SuperMicro 1U rackmount server chassis
 * Intel *P4 3.2GHz* Processor
 * *1GB PC3200* SDRAM (Single DIMM)
 * (2) Maxtor *80GB* SATA Hot Swap HDD's (Support 4 Hot Swap Drives)
 * Dual Onboard *Gigabit Ethernet*
 * Onboard CD-ROM, Video
 * 300W P/S
 * *PCI Riser Card supports Digium or Sangoma Interface Board*

Now if only they had added a dual hotswap power supply. I have searched
but haven't found a 1U box that has one. Anyone know a vendor that does
(barebones or just a 1U case with a dual hotswap PS)?

Regards,
Patrick  
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Re: [Asterisk-Users] asterisk log

[asterisk]

Thank you very much
I decided not lo lower the log information (leaving all  : full =
notice,warning,error,debug,verbose)

I started a weekly-rotation of the full log.

Andrea



   
 gincantalupo  
 [EMAIL PROTECTED] 
 software.com  To 
 Sent by:  Asterisk Users Mailing List -   
 asterisk-users-bo Non-Commercial Discussion   
 [EMAIL PROTECTED] asterisk-users@lists.digium.com   
 m.com  cc 
   
   Subject 
 12/10/2005 12.01  Re: [Asterisk-Users] asterisk log   
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




Hi,

[EMAIL PROTECTED] wrote:

Is there a way to

1) disable asterisk from writing in the full  log  ? (
/var/log/asterisk/full )


Take a look at /etc/asterisk/logger.conf

  or
2) implement a log rotation or similar of the full log ?

I see the full log grows a lot (about 100 MB per Month)


Use logrotate.

thanks in advance,

Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di
cancellarla.

Visitate il sito http://www.frameweb.it

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[Asterisk-Users] CHANNEL HANGUP ASSISTANCE

[yusuf]

Hi I have encountered a problem with my asterisk. Here is my set-up , I 
am using E1_PRI as signalling over a Nortel PABX. What i intended on 
doing is sending a call rejected signal . I have it set-up as 
PRI_CALLED=21 , it sends the signal but then it hangs up the channel , i 
need help sending the signal without hanging up the channel.


I have SET_VAR{PRI_CALLED=21} , and this does send the signal but i do 
not want it to hang up the channel. I am using asterisk version CVS 
19.07.2005 and the same version of ZAPTEL.


Any assistance regarding this issue is greatly appreciated THANX!!
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[Asterisk-Users] SNOM 360 Unknown SIP command 'PUBLISH'

[asterisk-Users]

Hi List



Im getting this notification from my one and
only SNOM 360 every time a number button is pushed.

I know that its only a notification, but it really
irritates me. Is it anything I can/should do anything about ??



Oct 12 10:34:33 NOTICE[3566]: chan_sip.c:10530
handle_request: Unknown SIP command 'PUBLISH' from '192.168.100.100'





By the way Im using * 1.0.9 CVS-HEAD September
15. 2005 



Best regards



BennyBad






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[Asterisk-Users] parameters documentation

[pellegrini]

Another trivial question:

Is there a place where all the parameters are documented ?
In example (my example!) I would like to know the meaning of a lot of
parameter that can be used in sip.conf,

A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060;
context=x) but other are not (at least for me)
i.e.:

type = peer, friend
insecure=very
host=dynamic

and so on.

At last, my need is:

Accept a non-registerd sip-strem from a well known ip address (and only
from that ip address)

I tried to add a

[testsip]
;username=testsip
type=friend
;secret=testsip
qualify=no
port=5060
nat=no
host=x.y.z.w
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=test  sip  testsip

that would work if the sip would be registered. But the SIP client is not
able to register.

I solved using the
context = from-sip-external ; Send unknown SIP callers to this context

and it works, but I have no more the control about who is sending me SIP
stream (anybody now can use my asterisk box...)

any help will be greatly appreciated

Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

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[Asterisk-Users] arcaplex / horizon isdn and analog multiplex

[Goran Skular]

Has anybody tried something like this:



http://www.arca-technologies.com/datasheets/arcaplexhorizon.pdf



It will be interesting to have ability to make systems like:



SCENARIO 1 (2 incoming BRI lines and 12 analog extensions  with ability
to connect additional isdn devices to s0 buses):



1 card with 8 BRI (from Junghanns or Beronet or someone else) ports (2
of them configured in TE mode and 6 of them in NT mode)



1 something that will convert e.g. 6 BRI to 12 analog FXS ports for
analog telephony equipment..





Or



SCENARIO 2 (1 incoming PRI E1 or T1 and 32 or more analog extensions)



2 Digium/Sangoma/Eicon whatsoever T1 or E1 cards (1 to telco, 1 to
something like this arcaplexhorizon)



1 Arcaplexhorizon ISDN and analog multiplexor with 32 analog ports (PO21/32A)







Maybe this Arcaplex can be used for 32 analog ports connected to
Asterisk with 1E1/T1 card 



Some thoutghts ?



Goran








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Re: [Asterisk-Users] supermicro with asterisk and tdm cards

[Steve Totaro]

 On Wed, 2005-10-12 at 00:59 -0400, Cory Andrews wrote:
 [snip]
  * SuperMicro 1U rackmount server chassis
  * Intel *P4 3.2GHz* Processor
  * *1GB PC3200* SDRAM (Single DIMM)
  * (2) Maxtor *80GB* SATA Hot Swap HDD's (Support 4 Hot Swap Drives)
  * Dual Onboard *Gigabit Ethernet*
  * Onboard CD-ROM, Video
  * 300W P/S
  * *PCI Riser Card supports Digium or Sangoma Interface Board*

 Now if only they had added a dual hotswap power supply. I have searched
 but haven't found a 1U box that has one. Anyone know a vendor that does
 (barebones or just a 1U case with a dual hotswap PS)?

 Regards,
 Patrick
 

My Sun server has hotswap power supplies.  I have not tried to install
asterisk on it since a couple years ago but I am pretty sure that I read
about * success on Solaris/SPARC servers.

Thanks,
Steve

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RE: [Asterisk-Users] help with broken voicemail

[Andy Goss]

drwxr-xr-x6 root root 1024 Oct 12 01:10 .
drwxr-xr-x6 root root 1024 Oct 12 01:15 ..
-rwxr-xr-x1 root root12801 Oct 11 21:28 busy.wav
drwxr-xr-x2 root root 1024 Oct 11 21:28 cust3
-rwxr-xr-x1 root root 3051 Oct 11 21:28 greet.wav
drwxr-xr-x2 root root 1024 Oct 11 21:28 inbox
drwxr-xr-x2 root root 1024 Oct 12 01:10 old
-rwxr-xr-x1 root root29895 Oct 11 21:28 unavail.wav
drwxr-xr-x2 root root 1024 Oct 11 21:28 work

-bash-2.05b# file unavail.wav
unavail.wav: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz




From: [EMAIL PROTECTED] on behalf of Tzafrir Cohen
Sent: Wed 10/12/2005 2:38 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] help with broken voicemail



On Tue, Oct 11, 2005 at 10:09:34PM -0400, Andy Goss wrote:
 Update -

 I made a backup of my entire voicemail directory then deleted it.  If I
 then try and record a greeting, it works.  Asterisk creates the folder
 structure and records the greeting.  If I try to copy the old file back
 into the directory, it wont work.  It's the same file name and
 everything.  The only thing I can figure might be an issue is that the
 voicemail drive is mounted as msdos so maybe there is something
 permissions different about the files that I cant see. 

 Any help would be appreciated. 

Please post the output of the following two commands:

ls -l /path/to/message.wav
file  /path/to/message.wav

Is it indeed a valid wav/RIFF file?

--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's 
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE: [Asterisk-Users] Areski Calling Card GUI

[Omar McKenzie]

I have check document , still not very clear on default html or php 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey
Sent: Wednesday, October 12, 2005 2:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Areski Calling Card GUI

If you haven't seen it already, this will be a lot of help to you.

http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Applicatio
n+The+idiots+guideV2

You should now be on step 12. :)


G


Omar McKenzie wrote:
 Hi
 
 I have gone thru the steps of installing AreskiCC, I would 
 like to know how to get access to the GUI interface of this application.
 
 
 
 
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Bob Goddard]

On Tuesday 11 Oct 2005 23:57, Lee Howard wrote:
 Bob Goddard wrote:
 On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
 Tom Rymes wrote:
 Use the right tool for the job!!!
 
 Use a hardware based DSP for faxing not software based.

 Why is a soft-DSP to be considered any less-capable than hardware ones?

Timing.

 The reason why I put IAXmodem together in the first place was because of
 a growing frustration that I've had with hardware chipsets and the lack
 of attention that the manufacturers generally afford to resolving
 fax-related bugs in their products.

Don't use any form of winmodem then.

[...]
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Re: [Asterisk-Users] E400P vs te410p vs te411p

[Steve Totaro]

 Hi,

 I found E400P quad PRI card quite cheap (749USD):


http://www.govarion.com/product_info.php?cPath=1products_id=2osCsid=68cdd6e3d08754

 in comparison to te410p (approx 1500 USD )


http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P

 Now newer generation with HW echo canceling emerged (te411p).

 I'm not sure in what things those two cards differ and what would be best
 option to buy (I believe there is big performance gap between them, but
 don't know how big and if it's worth of money) Also how do you find HW
 echo canceling in te411p ?

 Any advice, help ?

 Thanks in advance,

 regards,

 Rob.


Buy the TE410P that can be converted or upgraded to a TE411P for $100 less
than if you just purchased the TE411P straight up.  Test it as a TE410P and
if you dont have echo or can eliminate it then good, you make out well.  If
not you upgrade.

The E400P offloads most of its work to the system from what I understand so
you will use alot more system resources.  You also have no upgrade path.  If
you are lucky, you can use the E400P card if you have a beefy system already
that can handle it and you never need hardware echo cancellation, but thats
a gamble.

Thanks,
Steve Totaro

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Re: [Asterisk-Users] parameters documentation

[Steve Totaro]

www.voip-info.org

 Another trivial question:

 Is there a place where all the parameters are documented ?
 In example (my example!) I would like to know the meaning of a lot of
 parameter that can be used in sip.conf,

 A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060;
 context=x) but other are not (at least for me)
 i.e.:

 type = peer, friend
 insecure=very
 host=dynamic

 and so on.

 At last, my need is:

 Accept a non-registerd sip-strem from a well known ip address (and only
 from that ip address)

 I tried to add a

 [testsip]
 ;username=testsip
 type=friend
 ;secret=testsip
 qualify=no
 port=5060
 nat=no
 host=x.y.z.w
 dtmfmode=rfc2833
 context=from-internal
 canreinvite=no
 callerid=test  sip  testsip

 that would work if the sip would be registered. But the SIP client is not
 able to register.

 I solved using the
 context = from-sip-external ; Send unknown SIP callers to this context

 and it works, but I have no more the control about who is sending me SIP
 stream (anybody now can use my asterisk box...)

 any help will be greatly appreciated

 Andrea

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RE: [Asterisk-Users] error message when accessing voicemail

[Andy Goss]

-bash-2.05b# ls -la /var/spool/asterisk/voicemail/default/5933/
total 288
drwxr-xr-x6 root root32768 Oct 12 01:18 .
drwxr-xr-x   19 root root32768 Oct 12 01:17 ..
-rwxr-xr-x1 root root12936 Oct 12 01:14 busy.gsm
drwxr-xr-x2 root root32768 Oct 12 01:14 cust3
-rwxr-xr-x1 root root 3036 Oct 12 01:14 greet.gsm
drwxr-xr-x2 root root32768 Oct 12 07:54 inbox
drwxr-xr-x2 root root32768 Oct 12 02:01 old
-rwxr-xr-x1 root root30294 Oct 12 01:14 unavail.gsm
drwxr-xr-x2 root root32768 Oct 12 01:14 work

-bash-2.05b# ls -la /var/spool/asterisk/voicemail/default/5933/inbox/
total 128
drwxr-xr-x2 root root32768 Oct 12 07:54 .
drwxr-xr-x6 root root32768 Oct 12 01:18 ..
-rwxr-xr-x1 root root22110 Oct 12 07:54 msg.gsm
-rwxr-xr-x1 root root  264 Oct 12 07:54 msg.txt


Asterisk runs under root.  I fixed my other errors by converting the wav's to 
gsm, however they still dont make sense to me.  Any thoughts?

 
Thanks,
Andy



From: [EMAIL PROTECTED] on behalf of Tzafrir Cohen
Sent: Wed 10/12/2005 2:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] error message when accessing voicemail



On Wed, Oct 12, 2005 at 01:44:34AM -0400, Andy Goss wrote:
 If anyone could tell me what this error is all about, I would be very
 grateful. 

 Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
 path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not
 permitted
 Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
 path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not
 permitted

 Now, goodnight and thank you in advance

Under what user does Asterisk run?

ls -la /var/spool/asterisk/voicemail/default/5933

--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's 
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Modifying cmd VoicemailMain

[Nathan E. Pralle]

 1) Add words used in message retrieving guidance.
 I need to add different suffixes to numeric words due to Japanese way of
 mentioning time. (e.g. in English, you can say Five forty-five for 5:45,
 but in Japanese, we have to put hour and minute for respective time
 unit (meaning, VoicemailMain should pronounce as Five hours and forty-five
 minutes in Japanese). So, is there any way to add words modifying the
 regular word order?

You would have to edit the actual Asterisk source code and add in Japanese 
cases for all of the places where it chooses how to say each language.  
(Asterisk can say prompts in any language, but I don't think it has Japanese 
yet.)  It'd be a  custom job.

 2) Disable most of the key function guidance for retrieving the message.
 I don't want too much function guidance of VoicemailMain saying such as 3
 for advanced options and the like. I just want to hear just a few
 important keys to press. So, is there any way I can separately disable
 guidance for each key functions

Same idea as above.

Not tough if you're a C coder, but definately not a matter of a few flicks of 
a switch.

Nathan

-- 

Interesting things abide:
http://www.nathanpralle.com

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Re: [Asterisk-Users] supermicro with asterisk and tdm cards

[Cory Andrews]

Yeah I should have picked up on that, single PCI Riser in this one, so 1 
card.  I don't know of any 1U solution out there that would give you 3 
PCI slots to work with, I think you'll have to go to a 2U at least to 
achieve this.


Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
fax - 716.630.1548



Anton Krall wrote:


Hey Cory!

How many PCI does it support? Im looking for an option that can support 3 or
more TDM cards, the idea I have is to have model #'s for a server that can
handle 1 E1/T1 cards nicely and one # for a server that can handle 3 or more
TDM cards (without IRQ conflicts) so I can offer both TDM or E1 solutions.



|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Cory Andrews

|Sent: Tuesday, October 11, 2005 11:59 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] supermicro with asterisk and tdm cards
|
|AK - this is a nice configuration for use with Asterisk.  If 
|you want cheap, there are cheaper alternatives, but this 
|configuration works flawlessly in our experience and is 
|affordable.  The SuperMicro server model # I am referencing is 
|SYS-5013C-MTB and it is a current model, but it's barebones 
|you need to buy your proc, RAM, HDD's, etc , seperately.

|
|* SuperMicro 1U rackmount server chassis
|* Intel *P4 3.2GHz* Processor
|* *1GB PC3200* SDRAM (Single DIMM)
|* (2) Maxtor *80GB* SATA Hot Swap HDD's (Support 4 Hot Swap Drives)
|* Dual Onboard *Gigabit Ethernet*
|* Onboard CD-ROM, Video
|* 300W P/S
|* *PCI Riser Card supports Digium or Sangoma Interface Board*
|
|
|Cory Andrews
|Senior Partner
|+++
|VOIPSupply.com
|454 Sonwil Drive
|Buffalo, NY 14225
|+++
|voice - 716.630.1555 X22
|email - [EMAIL PROTECTED]
|fax - 716.630.1548
|
|
|
|Anton Krall wrote:
|
|Guys.
|
|Anybody using supermicro mobos and chassis with TDM cards?
|I would like to know which models are you using (mobos and 
|chassis and 
|also
|CPUs) and how many TDM cards have you been able to put in without 
|having IRQ issues like in other cases.

|
|Ive read supermicro servers play nice with asterisk but it is always 
|good to ask I guess.

|
|Thx!
|
|AK
|
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|
|  
|

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|

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[Asterisk-Users] Second Request for help: hardware requirements

[Zadikem, Travis]

Hello all.  I am new to Asterisk as well as this group so please excuse me for 
a bit as I learn the 
ropes of Asterisk.  Anyway,  I currently am using a pap2-na adapter with Teliax 
and Mesa Networks (my isp) and
was wondering what I will need to get Asterisk running correctly.  I am 
wondering what I will need in the machine besides
a NIC card to handle my home traffic.  I have only 3 phones in the house and am 
disconnected from the public
network (qwest) all together.  I will be using VOIP for both incoming and 
outgoing, but need the asterisk side
of things so that I can do digit translation (911 to our local emergency 
station).  Please help.

Thanks,
Travis
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[Asterisk-Users] Outgoing call: hangup after answer

[Goran Skular]

For your information.. if someone get in
the same trouble.. problem is solved, but not with the software



We just changed our BRI NT device with a
different one.. from now on it works very well



We had Elcon NT1+2a/b and now it is
replaced with Santis ISDN NT1+2ab



Here is pri debug:









-- Making new call for cr 143

 -- Requested transfer
capability: 0x00 - SPEECH

 Protocol Discriminator: Q.931
(8) len=22

 Call Ref: len= 1 (reference 15/0xF)
(Originator)

 Message type: SETUP (5)

 [04 03 80 90 a3]

 Bearer Capability (len= 5) [ Ext:
1 Q.931 Std: 0 Info transfer capability: Speech (0)


Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)


Ext: 1 User information layer 1: A-Law (35)

 [18 01 81]

 Channel ID (len= 3) [ Ext: 1
IntID: Implicit, Other Spare: 0, Preferred Dchan: 0


ChanSel: B1 channel


]

 [6c 05 21 80 32 30 30]

 Calling Number (len= 7) [ Ext:
0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1)


Presentation: Presentation permitted, user number not screened (0) '200' ]

 [70 01 c1]

 Called Number (len= 3) [ Ext: 1
TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1) '' ]

 -- Called g1/

 Protocol Discriminator: Q.931
(8) len=11

 Call Ref: len= 1 (reference 143/0x8F)
(Terminator)

 Message type: SETUP ACKNOWLEDGE (13)

 [18 01 89]

 Channel ID (len= 3) [ Ext: 1
IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0


ChanSel: B1 channel


]

 [1e 02 82 88]

 Progress Indicator (len= 4) [ Ext:
1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public
network serving the local user (2)


Ext: 1 Progress Description: Inband information or appropriate pattern
now available. (8) ]

-- Processing IE 24 (cs0, Channel
Identification)

-- Processing IE 30 (cs0, Progress
Indicator)

 Protocol Discriminator: Q.931
(8) len=8

 Call Ref: len= 1 (reference 15/0xF)
(Originator)

 Message type: INFORMATION (123)

 [70 02 c1 30]

 Called Number (len= 4) [ Ext: 1
TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1) '0' ]

 Protocol Discriminator: Q.931
(8) len=8

 Call Ref: len= 1 (reference 15/0xF)
(Originator)

 Message type: INFORMATION (123)

 [70 02 c1 39]







 Called Number (len= 4) [ Ext: 1
TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1) '0' ]

 Protocol Discriminator: Q.931
(8) len=4

 Call Ref: len= 1 (reference 143/0x8F)
(Terminator)

 Message type: CALL PROCEEDING (2)

 -- Zap/1-1 is making
progress passing it to SIP/200-7b76

 Protocol Discriminator: Q.931
(8) len=8

 Call Ref: len= 1 (reference 143/0x8F)
(Terminator)

 Message type: ALERTING (1)

 [1e 02 84 88]

 Progress Indicator (len= 4) [ Ext:
1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public
network serving the remote user (4)


Ext: 1 Progress Description: Inband information or appropriate pattern
now available. (8) ]

-- Processing IE 30 (cs0, Progress
Indicator)



n
Zap/1-1 is ringing





NEW_HANGUP DEBUG: Calling q931_hangup,
ourstate Call Delivered, peerstate Call Received

 Protocol Discriminator: Q.931
(8) len=8

 Call Ref: len= 1 (reference 15/0xF)
(Originator)

 Message type: DISCONNECT (69)

 [08 02 81 90]

 Cause (len= 4) [ Ext: 1 Coding:
CCITT (ITU) standard (0) 0: 0 Location: Private network serving the
local user (1)


Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]

 -- Hungup 'Zap/1-1'









Hi,











When we make an outgoing call on ISDN (zaphfc) with overlap dialing
we get immidiate hangup after answer. But when we place a full number before
dialing everything is ok. Any help appriciated!! Thanks











here is info with debug:















 [1e 02 84 88]
 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Public network serving the remote user (4)

Ext: 1 Progress Description: Inband information or appropriate pattern
now available. (8) ]
 [1e 02 84 82]
 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Public network serving the remote user (4)

Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ]
-- Processing IE 30 (cs0, Progress Indicator)
-- Processing IE 30 (cs0, Progress Indicator)












 -- Zap/1-1 is ringing, hanging up.


















NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call
Received
 Protocol Discriminator: Q.931 (8) len=8
 Call Ref: len= 1 (reference 64/0x40) (Originator)
 Message type: DISCONNECT (69)
 [08 02 81 90]
 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Private network serving the local user (1)

Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]











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[Asterisk-Users] unloading TE110P bristuffed module cause kernelpanic

[Francesco Angi]

 Same problem with debian sarge on a dell and asterisk 1.0.7 from 
 packages, unloading the module freezes the system, (rebooting the 
 machine worked right), I installed zaptel 1.2beta and it seems to
work, 
 but I haven't really tested it, only loaded/unloaded/loaded and
placed a 
 couple of calls.

Interesting. The zaptel part of the bristuff patch is rather small and
does not seem to have much to do with locks or with the init code, at
first glance.

Also note that the zaptel patch actually applies cleanly to 1.2 .
Contact me by email for packages. Though I suspect that it is not going
to solve the problem.


Patched zaptel 1.2.0beta1 with bristuff 0.2.0-RC8o and now modules load
and unload well.
Now have to try Asterisk, but this is another story...

Thanks for help,
_fangi_


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RE: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Tom Rymes]

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Bob Goddard
 Sent: Tuesday, October 11, 2005 6:35 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
 are fax-capable?

 On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
  Tom Rymes wrote:
   Frankly, I would recommend that you forget about trying
 to fax with
   Asterisk. Buy a good Multitech analog modem and install HylaFAX.
  
   Use the right tool for the job!!!
 
  Actually, you can use HylaFAX and Asterisk together.
 
https://sourceforge.net/projects/iaxmodem/
 
  Just be certain that your audio path doesn't run over any
 lossy medium
  (so run IAXmodem on your Asterisk box).

 I'll expand on what Tom meant

 Use a hardware based DSP for faxing not software based.

Actually Bob, that isn't what I meant. Lee simply suggested a different
way (IAXModem instead of analog modem) of implementing what I meant. I
would still recommend using analog if you can but, if you cannot, use
IAXModem from Lee.

Asterisk's faxing capabilities are not nearly as advanced, stable, or
easy to set up as HylaFAX. Also, there seem to be many problems with
frame slipping and the like that screw up faxing over Digium cards, and
maybe others as well.

Either way, I was just saying that grabbing a good modem (see HylaFAX
list archives for suggestions - NOT USRobotics!!!) and installing
HylaFAX would be easier, more reliable, and all-in-all, a better
solution than messing with Asterisk's built-in fax capability.

Tom



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[Asterisk-Users] X100P callerid ETSI - caller*ID failed checksum

[oner asterisk]

Dear All, 

I am a newbie about asterisk. I have 1x X100P card 3x Sip phone 

I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I gotno caller id, even my direct PSTN service operator. So at that moment I was using *
1.0.9. than I changed to [EMAIL PROTECTED] 1.3(1.0.9). I got same result. Then I connected an caller id analog phone to X100P phone port . I could see all CIDs from all operators (PSTN GSM) 


Isearched CID standards. Our country is using ETSI standards. But I am not sure about which method we are using ( FSK or DTMF)


Then trials and patch searches and I find out some but no result, even I lost CID completely. I setup system again and again. I am using Debian 
3.1 and as asterisk 
1-
 asterisk 1.0.9 with zaptel 1.0.9
2-
 asterisk 1.2 beta1 with zaptel 1.2.beta1
3-
 asterisk 1.2 betal1 with zaptel 1.0.9
4-
 asterisk 1.0.9 with 1.0.7
5-
 with CVS
and with all configs my My zaptel.conf is 

loadzone=us
defaultzone=us
fxsks=1 


Zapata.conf ( I played with commented parameters)

[channels] 
signalling=fxs_ks
context=incoming
language=us
immediate=no
usecallerid=yes
callerid=asreceived

useincomingcalleridonzaptransfer=yes ; I tried with /without / yes /no
usedistinctiveringdetection=no ; our op is not using
;cidsignalling=bell ; I tried bell, v23, dtmf
;cidstart=ring ; As I can see from my analog phone CID is coming after first ring. I tried ring and polarity


musiconhold=default

busydetect=no
;busycount=4
;busypattern=500,500


;hidecallerid=no
;callwaiting=yes
;usecallingpres=yes

;callwaitingcallerid=yes
;threewaycalling=yes 
;transfer=yes
;canpark=yes
cancallforward=yes 
;callreturn=yes
;echocancel=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
channel=1 

callgroup=1
pickupgroup=1


Error message is 
-- Starting simple switch on 'Zap/1-1'
-- Asterisk Urgent handler
callerid.c:312 callerid_feed: Caller*ID failed checksum

and usually with a ":success" at the end. Which make no sense


Now I am feeling like donkey but wanna solve this issue, So My Questions are: 
1-
 What parameters I should use for ETSI in Zapata.conf and zaptel.conf

2-
 How Can I distinguish FSK and DMTF types ? any test method or any software to be sure ?

3-
 I see clidtest program but no success to compile it . do you know how ?

4-
 I tested indication tones also it seems okay do you think is it part of the problem?


How can I solvecallerid issue 
Thanks

Cheers,
Oner
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RE: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Tom Rymes]

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 trixter http://www.0xdecafbad.com
 Sent: Tuesday, October 11, 2005 5:09 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
 are fax-capable?


 On Tue, 2005-10-11 at 17:01 -0400, Tom Rymes wrote:
  Frankly, I would recommend that you forget about trying to fax with
  Asterisk. Buy a good Multitech analog modem and install HylaFAX.
 
  Use the right tool for the job!!!

 Asterisk may be able to fax better in the somewhat near
 future.  One of the things holding up T.38 support is the
 inability for asterisk to switch codecs on the fly.  I am not
 saying that is the only thing, just one of the things.  Well
 1.2 is supposed to have better support in that regard, which
 means that work on T.38 can happen in a better way in the future.


This is good news. (I would like to be able to receive faxes reliably
over our PRI)

Until then, however, I still recommend HylaFAX.

Tom



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Re: [Asterisk-Users] parameters documentation

[asterisk]

Really strange answer. I am non used to search on playboy.com.

Anyway, if you try to search
insecure=very
on www.voip-info.org, you find 742 links , a bit more for me. (I just want
to know what it means)

Moreovere, the first 20 links are non accessible at all

http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecurediff=6

they speak about tiki-pagehistory.php, which appears not to exist.

no other comments about this.


I know about one project , asterisk documentation project

http://www.asteriskdocs.org

in its home page, the first line is





 Great software needs great documentation.  




I really hope this project will be implemented, without documentation
evrything is too hard

Andrea



   
 Steve Totaro
 [EMAIL PROTECTED] 
 echnologies.com   To 
 Sent by:  Asterisk Users Mailing List -  
 asterisk-users-bo Non-Commercial Discussion  
 [EMAIL PROTECTED] asterisk-users@lists.digium.com   
 m.com  cc 
   
   Subject 
 12/10/2005 14.53  Re: [Asterisk-Users] parameters 
   documentation   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   





www.voip-info.org

 Another trivial question:

 Is there a place where all the parameters are documented ?
 In example (my example!) I would like to know the meaning of a lot of
 parameter that can be used in sip.conf,

 A lot of these keywords are intuitive keywords (i.e.
NAT=YES/NO;PORT=5060;
 context=x) but other are not (at least for me)
 i.e.:

 type = peer, friend
 insecure=very
 host=dynamic

 and so on.

 At last, my need is:

 Accept a non-registerd sip-strem from a well known ip address (and only
 from that ip address)

 I tried to add a

 [testsip]
 ;username=testsip
 type=friend
 ;secret=testsip
 qualify=no
 port=5060
 nat=no
 host=x.y.z.w
 dtmfmode=rfc2833
 context=from-internal
 canreinvite=no
 callerid=test  sip  testsip

 that would work if the sip would be registered. But the SIP client is not
 able to register.

 I solved using the
 context = from-sip-external ; Send unknown SIP callers to this context

 and it works, but I have no more the control about who is sending me SIP
 stream (anybody now can use my asterisk box...)

 any help will be greatly appreciated

 Andrea

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[Asterisk-Users] Zaptel Debug: T1: Lost our place, resyncing

[Geoff Manning]

We are trying to debug a connection between Asterisk and a legacy PBX (Mitel
SX200). We turned on the Zaptel debugging and we get the following message
quite frequently:

Oct 12 07:14:09 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:09 localhost last message repeated 3 times
--
Oct 12 07:14:11 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:11 localhost last message repeated 7 times
--
Oct 12 07:14:15 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:15 localhost last message repeated 5 times
--
Oct 12 07:14:17 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:17 localhost last message repeated 7 times
--
Oct 12 07:14:26 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:26 localhost last message repeated 5 times
--
Oct 12 07:14:28 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:28 localhost last message repeated 6 times
--
Oct 12 07:14:33 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:33 localhost last message repeated 5 times
--
Oct 12 07:14:33 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:33 localhost last message repeated 7 times
--

What does this log entry mean?

We are using a Digium TE110P connecting via T1 crossover cable to a Mitel
SX200 T1 Card.  Our connection just recently began producing enough slip and
frame errors to cause the Mitel to automatically take it's T1 card offline.
Since we just started with the debugging, we're not sure if we would have
had these entries when the system was operational.

We are using an identical server/TE110P integrating with an IsoTec Executone
and we do not receive those log entries.

Thanks,
Geoff

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Re: [Asterisk-Users] parameters documentation

[asterisk]

 I really hope this project will be implemented, without documentation
evrything is too hard

Not for the thousands of people that have figured it out.

3Com NBX might be more your speed and plenty of documentation.



 Really strange answer. I am non used to search on playboy.com.

 Anyway, if you try to search
 insecure=very
 on www.voip-info.org, you find 742 links , a bit more for me. (I just want
 to know what it means)

 Moreovere, the first 20 links are non accessible at all


http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecurediff=6

 they speak about tiki-pagehistory.php, which appears not to exist.

 no other comments about this.
 

 I know about one project , asterisk documentation project

 http://www.asteriskdocs.org

 in its home page, the first line is





  Great software needs great documentation.




 I really hope this project will be implemented, without documentation
 evrything is too hard

 Andrea




  Steve Totaro
  [EMAIL PROTECTED]
  echnologies.com   To
  Sent by:  Asterisk Users Mailing List -
  asterisk-users-bo Non-Commercial Discussion
  [EMAIL PROTECTED] asterisk-users@lists.digium.com
  m.com  cc

Subject
  12/10/2005 14.53  Re: [Asterisk-Users] parameters
documentation

  Please respond to
   Asterisk Users
   Mailing List -
   Non-Commercial
 Discussion
  [EMAIL PROTECTED]
  ists.digium.com







 www.voip-info.org

  Another trivial question:
 
  Is there a place where all the parameters are documented ?
  In example (my example!) I would like to know the meaning of a lot of
  parameter that can be used in sip.conf,
 
  A lot of these keywords are intuitive keywords (i.e.
 NAT=YES/NO;PORT=5060;
  context=x) but other are not (at least for me)
  i.e.:
 
  type = peer, friend
  insecure=very
  host=dynamic
 
  and so on.
 
  At last, my need is:
 
  Accept a non-registerd sip-strem from a well known ip address (and only
  from that ip address)
 
  I tried to add a
 
  [testsip]
  ;username=testsip
  type=friend
  ;secret=testsip
  qualify=no
  port=5060
  nat=no
  host=x.y.z.w
  dtmfmode=rfc2833
  context=from-internal
  canreinvite=no
  callerid=test  sip  testsip
 
  that would work if the sip would be registered. But the SIP client is
not
  able to register.
 
  I solved using the
  context = from-sip-external ; Send unknown SIP callers to this context
 
  and it works, but I have no more the control about who is sending me SIP
  stream (anybody now can use my asterisk box...)
 
  any help will be greatly appreciated
 
  Andrea

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Re: [Asterisk-Users] MWI for endpoints not registered at Asterisk

[Peter Bowyer]

On 12/10/05, Stojan Sljivic - Pamet [EMAIL PROTECTED] wrote:
 Hi,

 We have phones registered at another soft switch, and would like to use
 Asterisk as a Voicemail system.
 Is it possible and how to configure Asterisk to send NOTIFY messages (for
 MWI) to the endpoints that are not registered to the Asterisk?

I couldn't find a way round this, and ended up using a 'spare' line
presentation on my GXP-2000 phones to register to the voicemail server
simply to pick up the NOTIFYs. Since the phone only has a single MWI
LED, it doesn't matter which line the NOTIFY comes in on.

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: R: [Asterisk-Users] Problem setting up TDM22B card

[Lyle Giese]

I have not seen the output of modprob zaptel in this thread, which has
to take place before loading the other kernel drivers.

Lyle


so
mesh s wrote:

  Hi,

I changed the mother board (MB) but it is giving still
the same problem.
  
  

  

  

  

  ouput of dmesg|tail 
  

  

  

  

  
  f6 != 58
f7 != 59
f8 != 58
f9 != 59
fa != 58
fb != 59
fc != 58
fd != 59
fe != 58
Freshmaker failed register test
  
  and I have also configured zaptel.conf correctly.

Whatz next? Can I assume that it is a hardware
problem?

Regards,
Somesh S. Shanbhag


--- John Novack [EMAIL PROTECTED] wrote:

  
  

somesh s wrote:



  Hi,

I didn't get any solution in the mailing list.
  

[http://asterisk.linkx.net/asteriskusers/200409/msg01167]


  What should be the next step?

Changing the machine???
Is it machine dependent?...

Regards,
Somesh S. Shanbhag

 

  

Have you talked with Digium support?

Their answer almost always is:

"Try another Motherboard"
They won't supply a list that is known to work, only
ones that are known 
NOT to work.
 From my limited experience, even if the MB says it
is PCI 2.2, the TDM 
card may or may not work.

If you don't want to change machines, then  use an
ATA or two Sipura's 
work great.

John Novack


  




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Re: [Asterisk-Users] supermicro with asterisk and tdm cards

[Patrick]

On Wed, 2005-10-12 at 08:42 -0400, Steve Totaro wrote:
 
  On Wed, 2005-10-12 at 00:59 -0400, Cory Andrews wrote:
  [snip]
   * SuperMicro 1U rackmount server chassis
   * Intel *P4 3.2GHz* Processor
   * *1GB PC3200* SDRAM (Single DIMM)
   * (2) Maxtor *80GB* SATA Hot Swap HDD's (Support 4 Hot Swap Drives)
   * Dual Onboard *Gigabit Ethernet*
   * Onboard CD-ROM, Video
   * 300W P/S
   * *PCI Riser Card supports Digium or Sangoma Interface Board*
 
  Now if only they had added a dual hotswap power supply. I have searched
  but haven't found a 1U box that has one. Anyone know a vendor that does
  (barebones or just a 1U case with a dual hotswap PS)?
 
  Regards,
  Patrick
  
 
 My Sun server has hotswap power supplies.  I have not tried to install
 asterisk on it since a couple years ago but I am pretty sure that I read
 about * success on Solaris/SPARC servers.

Thanks for the tip. I checked out the Sun site and the V4100 1U server
has a dual hotswap PS. I also noticed that it has 2 PCI-X slots. That
would be quite a space efficient box for 8 PRI's.

Regards,
Patrick

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Re: [Asterisk-Users] parameters documentation

[asterisk]

I come from a NBX100
No documentation available.
1 day it starts saying: syslog full and voicemail stop working
No one was able to tell me what was the meaning of that alert
.
3COM NBX anyway is a good product, but the price is too high, especially 4
years ago, and especially the price of the telephone is very high.

Andrea




   
 asterisk
 [EMAIL PROTECTED] 
 echnologies.com   To 
 Sent by:  Asterisk Users Mailing List -  
 asterisk-users-bo Non-Commercial Discussion  
 [EMAIL PROTECTED] asterisk-users@lists.digium.com   
 m.com  cc 
   
   Subject 
 13/10/2005 16.13  Re: [Asterisk-Users] parameters 
   documentation   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




 I really hope this project will be implemented, without documentation
evrything is too hard

Not for the thousands of people that have figured it out.

3Com NBX might be more your speed and plenty of documentation.



 Really strange answer. I am non used to search on playboy.com.

 Anyway, if you try to search
 insecure=very
 on www.voip-info.org, you find 742 links , a bit more for me. (I just
want
 to know what it means)

 Moreovere, the first 20 links are non accessible at all


http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecurediff=6


 they speak about tiki-pagehistory.php, which appears not to exist.

 no other comments about this.
 

 I know about one project , asterisk documentation project

 http://www.asteriskdocs.org

 in its home page, the first line is





  Great software needs great documentation.




 I really hope this project will be implemented, without documentation
 evrything is too hard

 Andrea




  Steve Totaro
  [EMAIL PROTECTED]
  echnologies.com
To
  Sent by:  Asterisk Users Mailing List -
  asterisk-users-bo Non-Commercial Discussion
  [EMAIL PROTECTED] asterisk-users@lists.digium.com
  m.com
cc


Subject
  12/10/2005 14.53  Re: [Asterisk-Users] parameters
documentation

  Please respond to
   Asterisk Users
   Mailing List -
   Non-Commercial
 Discussion
  [EMAIL PROTECTED]
  ists.digium.com







 www.voip-info.org

  Another trivial question:
 
  Is there a place where all the parameters are documented ?
  In example (my example!) I would like to know the meaning of a lot of
  parameter that can be used in sip.conf,
 
  A lot of these keywords are intuitive keywords (i.e.
 NAT=YES/NO;PORT=5060;
  context=x) but other are not (at least for me)
  i.e.:
 
  type = peer, friend
  insecure=very
  host=dynamic
 
  and so on.
 
  At last, my need is:
 
  Accept a non-registerd sip-strem from a well known ip address (and only
  from that ip address)
 
  I tried to add a
 
  [testsip]
  ;username=testsip
  type=friend
  ;secret=testsip
  qualify=no
  port=5060
  nat=no
  host=x.y.z.w
  dtmfmode=rfc2833
  context=from-internal
  canreinvite=no
  callerid=test  sip  testsip
 
  that would work if the sip would be registered. But the SIP client is
not
  able to register.
 
  I solved using the
  context = from-sip-external ; Send unknown SIP callers to this context
 
  and it works, but I have no more the control about who is sending me
SIP
  stream (anybody now can use my asterisk box...)
 
  any help will be greatly appreciated
 
  Andrea

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[Asterisk-Users] Re: parameters documentation

[Doug Meredith]

[EMAIL PROTECTED] wrote:

Anyway, if you try to search
insecure=very
on www.voip-info.org, you find 742 links , a bit more for me. (I just want
to know what it means)

I think the search is broken there.  Just go in under Asterisk and
look for where the configuration files are documented.

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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Re: [Asterisk-Users] Re: parameters documentation

[asterisk]

Thank you very much for your answer.
I searched the wiki using your criteria, and I found

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

which seems to be the answer to my question

thank you again

Andrea




   
 Doug Meredith 
 [EMAIL PROTECTED] 
 yridge.comTo 
 Sent by:  asterisk-users@lists.digium.com 
 asterisk-users-bo  cc 
 [EMAIL PROTECTED] 
 m.com Subject 
   [Asterisk-Users] Re: parameters 
   documentation   
 12/10/2005 11.31  
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




[EMAIL PROTECTED] wrote:

Anyway, if you try to search
insecure=very
on www.voip-info.org, you find 742 links , a bit more for me. (I just want
to know what it means)

I think the search is broken there.  Just go in under Asterisk and
look for where the configuration files are documented.

Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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Re: [Asterisk-Users] nat and wandering phones

[bails]

Hey that works great, only problem is how do i configure an outbound 
i.e. from asterisk to firefly extension?

Is this possible?

Cheers

Bails



Michael Graves wrote:


I do this all the time. I simply use the Firefly IAX2 soft phone and
don't bother with SIP at all. I forward port 4569 to my * box and it
just works.

Michael Graves

On Tue, 11 Oct 2005 14:30:36 +0100, bails wrote:

 


Hi all I'm looking for a solution to this problem.

*boxinternet---nat---softphone

We have potential customers who will be travelling the world with 
laptops/pda's.
They need to be able to connect to the asterisk box via ip wherever they 
are and will have no control over nat whatsoever.


I have read that STUN offers this service, but cannot picture in my mind 
how this works, especially with no port forwarding on the nat (I mean, 
how the hell do packets traverse the firewall to find the end device).


Any suggestions welcome

Cheers Bails
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--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



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Re: [Asterisk-Users] parameters documentation

[Steve Totaro]

There is plenty of documentation online for both the 3com and *.  You have
to have good search skills I guess.

3com has the best knowledge base I have seen.
http://knowledgebase.3com.com/ and there are tons of 3com dealers that can
help.

I think you may need to learn some basic networking before learning
asterisk.  NAT is a very basic concept in networking as well as ports such
as 5060 (standard port for SIP).

There is a very steep learning curve for asterisk and networking in general.
If you want to learn it then you need to dig into the wiki and read all the
posts that come across the user's list (well maybe not all  of them).

There are plenty of consultants that you can hire if you are not up to it.


- Original Message - 
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Wednesday, October 12, 2005 10:34 AM
Subject: Re: [Asterisk-Users] parameters documentation


 I come from a NBX100
 No documentation available.
 1 day it starts saying: syslog full and voicemail stop working
 No one was able to tell me what was the meaning of that alert
 .
 3COM NBX anyway is a good product, but the price is too high, especially 4
 years ago, and especially the price of the telephone is very high.

 Andrea





  asterisk
  [EMAIL PROTECTED]
  echnologies.com   To
  Sent by:  Asterisk Users Mailing List -
  asterisk-users-bo Non-Commercial Discussion
  [EMAIL PROTECTED] asterisk-users@lists.digium.com
  m.com  cc

Subject
  13/10/2005 16.13  Re: [Asterisk-Users] parameters
documentation

  Please respond to
   Asterisk Users
   Mailing List -
   Non-Commercial
 Discussion
  [EMAIL PROTECTED]
  ists.digium.com






  I really hope this project will be implemented, without documentation
 evrything is too hard

 Not for the thousands of people that have figured it out.

 3Com NBX might be more your speed and plenty of documentation.



  Really strange answer. I am non used to search on playboy.com.
 
  Anyway, if you try to search
  insecure=very
  on www.voip-info.org, you find 742 links , a bit more for me. (I just
 want
  to know what it means)
 
  Moreovere, the first 20 links are non accessible at all
 
 

http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecurediff=6

 
  they speak about tiki-pagehistory.php, which appears not to exist.
 
  no other comments about this.
  
 
  I know about one project , asterisk documentation project
 
  http://www.asteriskdocs.org
 
  in its home page, the first line is
 
 
 
 
 
   Great software needs great documentation.
 
 
 
 
  I really hope this project will be implemented, without documentation
  evrything is too hard
 
  Andrea
 
 
 
 
   Steve Totaro
   [EMAIL PROTECTED]
   echnologies.com
 To
   Sent by:  Asterisk Users Mailing List -
   asterisk-users-bo Non-Commercial Discussion
   [EMAIL PROTECTED] asterisk-users@lists.digium.com
   m.com
 cc
 
 
 Subject
   12/10/2005 14.53  Re: [Asterisk-Users] parameters
 documentation
 
   Please respond to
Asterisk Users
Mailing List -
Non-Commercial
  Discussion
   [EMAIL PROTECTED]
   ists.digium.com
 
 
 
 
 
 
 
  www.voip-info.org
 
   Another trivial question:
  
   Is there a place where all the parameters are documented ?
   In example (my example!) I would like to know the meaning of a lot of
   parameter that can be used in sip.conf,
  
   A lot of these keywords are intuitive keywords (i.e.
  NAT=YES/NO;PORT=5060;
   context=x) but other are not (at least for me)
   i.e.:
  
   type = peer, friend
   insecure=very
   host=dynamic
  
   and so on.
  
   At last, my need is:
  
   Accept a non-registerd sip-strem from a well known ip address (and
only
   from that ip address)
  
   I tried to add a
  
   [testsip]
   ;username=testsip
   type=friend
   ;secret=testsip
   qualify=no
   port=5060
   nat=no
   host=x.y.z.w
   dtmfmode=rfc2833
   context=from-internal
   canreinvite=no
   callerid=test  sip  testsip
  
   that would work if the sip would be registered. But the SIP client is
 not
   able to register.
  
   I solved using the
   context = 

[Asterisk-Users] Patton SmartNode

[[EMAIL PROTECTED]]

Does anybody have any experience using a Patton SmartNode as a SIP/Telco 
gateway with Asterisk?  They seem really inexpensive and appear to 
support all of the necessary features, but I don't have any experience 
with their products, so I don't know if they are any good.  We are 
currently using a Cisco 2600 w/ PRI card and it works fine, but I was 
looking for someone else as a possible alternative.  Thanks.


Peder
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Lee Howard]

Tom Rymes wrote:


(I would like to be able to receive faxes reliably
over our PRI)

Until then, however, I still recommend HylaFAX.
 



If your PRI comes in to a TE405P or somesuch then you can pass fax DIDs 
out through another port on the TE405P and out to a T1 faxmodem (such as 
a Patton 2977) or a T1 channel bank and then to analog modems.


Lee.
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[Asterisk-Users] Calibrating both RX and TX gain?

[tmassey]

Hello!

I'm having an echo problem with a TDM
card. The TDM card is being fed by a channel bank just 12 or so feet
away. When you put an analog handset on the line, both the RX and
TX volume seem to be just fine. However, when I use the TDM card,
I have to have an rxgain of 13.5, and even then, the audio is relatively
quiet. I'm also getting echo on these lines, so I have turned the
txgain down as low as I can and still be heard. Right now, it's at
-6, but it will have to come up some because that is too quiet. But
I still have echo.

I am in the middle of trying to get
a milliwatt test line to calibrate the rxgain properly. However,
this won't help me with the txgain, will it? How can I properly calibrate
the txgain? By ear? Or is there a more scientific method?

For example, once I have the rxgain
calibrated for all of the lines, could I then call into, say, Zap/3 from
Zap/4 and run Milliwatt() on Zap/3 and use ztmonitor on Zap/4 to calibrate
it? I'm sure it's not perfect, but would it be close enough?

A second question: doesn't it
seem wrong that my rxgain and txgain are so far off when I'm just talking
to a channel bank 12 feet away? I sure don't have cable loss. It
sure seems like the impedance is way off or something. Is there a
way to test this further, rather than just cranking up the gain? My
guess is that using the milliwatt line will just tell me to make the rxgain
higher, which will probably just make the echo issues worse... 

Tim Massey
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[Asterisk-Users] Sangoma FXO/FXS cards?

[Paul Dugas]

Saw an ad in the latest Linux Journal for Sangoma's FXO/FXS analog cards. 
it was part of an ad for a reseller.  I can't find anything on the
resellers site or Sangoma's site either.  Did the ad jump the gun or
someting?  Is this for real?

Paul
-- 
Paul Dugas, Computer Engineer   Dugas Enterprises, LLC
[EMAIL PROTECTED] phone: 404-932-1355   522 Black Canyon Park
http://dugas.cc fax: 866-751-6494   Canton, GA 30114 USA
--
Onsite at GDOT W.Annex 404-463-2860 x199
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[Asterisk-Users] Polycom: Button Remapping, HELP!

[Matthew T. O'Connor]

I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional but 
it speeds it up.) to park a call.  Personally I think this is easy, but 
my users would like one button to do this for them.  The Polycom has 
buttons we don't need (Transfer  Services), it would be nice if I could 
remap one of those buttons to dial #70#.  Or if I could add a soft 
button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Mojo with Horan Company, LLC]

Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
itemfnPark/fnct#70#/ctsd3/sd/item) and then in ipmid.cfg:
keys key.IP_500.31.function.prim=SpeedDial 
key.IP_500.31.subPoint.prim=3/


This tells the phone to run Speed Dial 3 whenever the Services button 
(button #31 on a 500/501) is pressed.  I hope someone can help us 
configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:
I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional but 
it speeds it up.) to park a call.  Personally I think this is easy, but 
my users would like one button to do this for them.  The Polycom has 
buttons we don't need (Transfer  Services), it would be nice if I could 
remap one of those buttons to dial #70#.  Or if I could add a soft 
button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] Sangoma FXO/FXS cards?

[Nathan C. Smith]

They will be announced formally soon.

-Original Message-
From: Paul Dugas [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, October 12, 2005 10:41 AM
To: Asterisk Mailing List
Subject: [Asterisk-Users] Sangoma FXO/FXS cards?


Saw an ad in the latest Linux Journal for Sangoma's FXO/FXS analog cards. 
it was part of an ad for a reseller.  I can't find anything on the resellers
site or Sangoma's site either.  Did the ad jump the gun or someting?  Is
this for real?

Paul
-- 
Paul Dugas, Computer Engineer   Dugas Enterprises, LLC
[EMAIL PROTECTED] phone: 404-932-1355   522 Black Canyon Park
http://dugas.cc fax: 866-751-6494   Canton, GA 30114 USA
--
Onsite at GDOT W.Annex 404-463-2860 x199
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[Asterisk-Users] Problem with PRI and Ericsson AXE 10

[Juan Manuel Coronado Z.]

Hi everyone,

I have a PRI conection on an * system running Asterisk 1.0.9, libpri 1.0.9 and 
zaptel 1.0.9.2 connected to an AXE 10 (APZ 21220 System 64) in the network 
side. I know the system and the wildcard I´m using are ok because I´ve used 
them before with other PRI connections (to a Siemens EWSD) without any problem.

First the PRI didn´t work (I got the TE110P alarmed with red). After 
troubleshooting with the Telco, they told me to disable crc4 and then I got the 
green ligth, the 30 B channels cleared, and I was able to dialout to the PSTN 
but only until the channels restart, because this made the call to hangup. 
Besides, when I make a call towards the system, I got a busy tone from the PSTN 
and no activity on the CLI.

The Telco has called to the system too and they say they got a connection 
reject even they see the channels available.

For troubleshooting porpouses, they have enabled the most simple configuration 
of PBX in the network side of the PRI (no DIDs, 30 B channels and one header).

These are the messages that I got in the CLI:

051008-153001 WARNING[27660]: chan_zap.c:7182 zt_pri_error: PRI: Short write: 
-1/15 (Unknown error 500)
051008-153001 WARNING[27660]: chan_zap.c:3195 zt_handle_event: Detected alarm 
on channel 1: Yellow Alarm
-- Hungup 'Zap/1-1'
  == No one is available to answer at this time
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 2: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 3: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 4: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 5: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 6: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 7: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 8: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 9: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 10: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 11: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 12: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 13: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 14: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 15: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 17: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 18: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 19: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 20: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 21: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 22: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 23: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 24: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 25: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 26: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 27: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 28: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 29: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 30: Yellow Alarm
051008-153001 WARNING[27660]: chan_zap.c:5684 handle_init_event: Detected alarm 
on channel 31: Yellow Alarm
051008-153001 NOTICE[27660]: chan_zap.c:7428 pri_dchannel: PRI got event: Alarm 
(4) on Primary D-channel of span 1
051008-153001 WARNING[27660]: chan_zap.c:1938 pri_find_dchan: No D-channels 
available!  Using Primary on channel anyway 16!
051008-153001 NOTICE[27660]: chan_zap.c:5679 handle_init_event: Alarm cleared 
on channel 1
051008-153001 NOTICE[27660]: chan_zap.c:5679 handle_init_event: Alarm cleared 
on channel 2
051008-153001 NOTICE[27660]: chan_zap.c:5679 handle_init_event: 

RE: [Asterisk-Users] Sangoma FXO/FXS cards?

[Mohammad Shokuie]

Dear folk,
You are right, seems sangoma is going to produce FXO/FXS cards but its still in the lab and not released yet but will do it in near future.
Regards,M. Shokuie Nia,CEO,SENA Co.



From:"Nathan C. Smith" [EMAIL PROTECTED]Reply-To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo:'Asterisk Users Mailing List - Non-CommercialDiscussion' asterisk-users@lists.digium.comSubject:RE: [Asterisk-Users] Sangoma FXO/FXS cards?Date:Wed, 12 Oct 2005 11:16:53 -0500MIME-Version:1.0Received:from lists.digium.com ([69.16.138.164]) by mc9-f5.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Wed, 12 Oct 2005 09:19:05 -0700Received:from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 4DD183FD335;Wed, 12 Oct 2005 11:16:54 -0500 (CDT)Received:from psmtp.com (exprod5mx132.postini.com 
[64.18.0.46])by lists.digium.com (Postfix) with SMTP id 821D23FD32Ffor asterisk-users@lists.digium.com;Wed, 12 Oct 2005 11:16:49 -0500 (CDT)Received:from source ([216.81.229.215]) by exprod5mx132.postini.com([64.18.4.10]) with SMTP; Wed, 12 Oct 2005 11:16:55 CDTReceived:from [10.1.1.2] ([10.1.1.2]:35084 "EHLO dsmexch.ipmvs.com")by mail.ipmvs.com with ESMTP id S53094AbVJLQQz (ORCPTrfc822;asterisk-users@lists.digium.com);Wed, 12 Oct 2005 11:16:55 -0500Received:by DSMEXCH with Internet Mail Service (5.5.2653.19)id RHY6CK8W; Wed, 12 Oct 2005 11:16:55 -0500They will be announced formally soon.-Original Message-From: Paul Dugas [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 12, 2005 10:41 AMTo: Asterisk Mailing ListSubject: 
[Asterisk-Users] Sangoma FXO/FXS cards?Saw an ad in the latest Linux Journal for Sangoma's FXO/FXS analog cards.it was part of an ad for a reseller.I can't find anything on the resellerssite or Sangoma's site either.Did the ad jump the gun or someting?Isthis for real?Paul--Paul Dugas, Computer Engineer Dugas Enterprises, LLC[EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Parkhttp://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA--Onsite at GDOT W.Annex 404-463-2860 x199___--Bandwidth and Colocation sponsored by Easynews.com 
--Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersDon't just search. Find. MSN Search Check out the new MSN Search!

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[Asterisk-Users] AGI and set_callerid for number and name

[Serge Lhermitte]

Hi,


I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber. 

Is it at at possible ?

Cheers.
SL


 
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[Asterisk-Users] sound very loud (saturated) through IAX2 and SIP

[Goran]

I have very loud sound through IAX2 and SIP channels, even very
saturated in some moments.

Why? How to change sound level (on IAX2 and SIP channels)?

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RE: [Asterisk-Users] AGI and set_callerid for number and name

[Pedro Nunes]

Curse,

Look at this php script ...

Contactlookup.agi

#!/usr/local/bin/php -q
 ?php
 ob_implicit_flush(true);
 set_time_limit(6);
 $in = fopen(php://stdin,r);
 $stdlog = fopen(/var/log/asterisk/my_agi.log, w);

 // toggle debugging output (more verbose)
 $debug = true;

 // Do function definitions before we start the main loop
 function read() {
   global $in, $debug, $stdlog;
   $input = str_replace(\n, , fgets($in, 4096));
   if ($debug) fputs($stdlog, read: $input\n);
   return $input;
 }

 function errlog($line) {
   global $err;
   echo VERBOSE \$line\\n;
 }

 function write($line) {
   global $debug, $stdlog;
   if ($debug) fputs($stdlog, write: $line\n);
   echo $line.\n;
 }

 // parse agi headers into array
 while ($env=read()) {
   $s = split(: ,$env);
   $agi[str_replace(agi_,,$s[0])] = trim($s[1]);
   if (($env == ) || ($env == \n)) {
 break;
   }
 }

 // main program
 echo VERBOSE \Here we go!\ 2\n;
 read();
 $session = mssql_connect('mssql server' , 'username' , 'password' );
$result = mssql_query(select * from ContactDB WHERE
extension=.$agi['callerid'],$session );
 $row = mssql_fetch_array($result);
 mssql_close($session);
 if ($row['Name'] == ){
  write('SET VARIABLE NAME Not Found');
  read();
 } else {
  write('SET VARIABLE NAME '.$row['Name'].'');
  read();
 }
 fclose($in);
 fclose($stdlog);


And in extensions.conf

[extensions]
exten = 4501,1,agi,contactlookup.agi
exten = 4501,2,SetCIDName(${Name})
exten = 4501,3,Dial(SIP/421,15)


It looks to an mssql DB, try to find the callerID number in table
extensions, and then sets a variable named Name to the value of
table Name. Cool hah...



Pedro Nunes



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Serge
Lhermitte
Sent: quarta-feira, 12 de Outubro de 2005 17:57
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AGI and set_callerid for number and name


Hi,


I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber. 

Is it at at possible ?

Cheers.
SL


 
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RE: [Asterisk-Users] Real Life FAX sending receiving

[Jenna Cole]

thanx, i did it.
i just instaled the debian packages for asterisk and
asterisk-app-fax and its working this way

fax--fxs--ipnetwork--fxs--fax

obviously the fxs ports are digium cards in linux
machines running asterisk using sip. the fax quality
is perfect

now i am trying with the following scenario:

fax--fxs--ipnetwork--FXO--pbx--fax

when i transmit a fax, an 80 percent of the fax is OK
but there are lines that cannot be read because the
quality is bad. sometimes the lines overlap each
other, and sometimes the height of the line is smaller

Can anyone help me?

Thanx 

 --- [EMAIL PROTECTED] escribió:

 I can send/receive just fine on an eicon bri to a
 zaptel analog
 interface.
 
 I would say, if you wish to use faxing on a regular
 basis to a remote
 proxy though, you're possibly better off with a
 landline.
 
 Regards,
 Greg 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Doug Lytle
 Sent: Monday, October 03, 2005 3:23 PM
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [Asterisk-Users] Real Life FAX sending
 receiving
 
 Jenna Cole wrote:
 
 receive the fax via SIP and send it to my
 faxmachine.
 I also want to send a fax from my faxmachine
 through the digium card, 
 so asterisk should send the fax via SIP to the
 gateway, which also has 
 a faxmachine connected.
 
 is this possible?
   
 
 Short answer, no.  Long answer can be found here:
 
 http://www.soft-switch.org/spandsp_faq/ar01s04.html
 
 Doug
 
 -- 
  
 Ben Franklin quote:
 
 Those who give up essential liberties for temporary
 safety deserve
 neither liberty nor safety.
 
 
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Re: [Asterisk-Users] Calibrating both RX and TX gain?

[Mojo with Horan Company, LLC]

Hello :)

For example, once I have the rxgain calibrated for all of the lines, 
could I then call into, say, Zap/3 from Zap/4 and run Milliwatt() on 
Zap/3 and use ztmonitor on Zap/4 to calibrate it?  I'm sure it's not 
perfect, but would it be close enough?
That's exactly what you do.  Once I had adjusted my rxgains to calibrate 
them to the signal that the phoneco gave me, I just dialed out of one 
line and into another.  Everything's supposedly digital on the phoneco 
side, so no loss should occur.  (because with the rxgains you've already 
compensated for what will happen on the inbound trip through the 
copper).  So by then adjusting your txgains on each channel, you can 
feel confident that the phoneco is accurately representing to you how 
you sound from its point of view.


A second question:  doesn't it seem wrong that my rxgain and txgain are 
so far off when I'm just talking to a channel bank 12 feet away?  I sure 
don't have cable loss.  It sure seems like the impedance is way off or 
something.  Is there a way to test this further, rather than just 
cranking up the gain?  My guess is that using the milliwatt line will 
just tell me to make the rxgain higher, which will probably just make 
the echo issues worse...
It does seem like something else is wrong.  You shouldn't require such 
high rxgains in my opinion, but I have no idea what could be causing 
this need.


Mojo
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Re: [Asterisk-Users] Calibrating both RX and TX gain?

[Shaw Terwilliger]

On Wed, Oct 12, 2005 at 12:05:32PM -0400, [EMAIL PROTECTED] wrote:
 I am in the middle of trying to get a milliwatt test line to calibrate the 
 rxgain properly.  However, this won't help me with the txgain, will it? 
 How can I properly calibrate the txgain?  By ear?  Or is there a more 
 scientific method?

Maybe I can help.

I had a similar problem.  I was using a Digium TE205P card and two
Rhino channel banks, and every call that was bridged from a phone on
an FXS interface to a PSTN line on an FXO interface was (1) loud and
(2) had an echo with a tiny delay (maybe 30ms).  The echo sounded almost
like excess sidetone, but was delayed enough to phase shift the speech
and make things sound hollow.  I could verify that what was being
transmitted was coming back on the RX channel of the PSTN interface
(using ztmonitor).  I'm using Nortel analog, wall-powered phones (pretty
nice models).

I had echo cancellation on, and had tried all possible configuration 
settings for taps, etc.  Nothing killed my echo.

I had tried adjusting all the gains down in Asterisk for all the interfaces,
but that didn't work.

I contacted Rhino to see if they had any suggestions, and they were
able to give me a few.  What finally worked was setting the Asterisk gains
back to 0 for all channels, then adjusting the gains down on the channel banks
themselves for the phone (FXS) interfaces only.  A huge improvement!  My
current adjustements are the following:

On the Rhino channel banks:

  For FXS (phones) interfaces:

rx -4 dB
tx -4 dB

  For FXO (PSTN lines) interfaces:

rx 0 dB (default)
tx 0 dB (default)

In Asterisk's zaptel.conf:

  context=phones
  rxgain=3.0; This is to compensate for the drop in volume because of
; the -4 dB setting on the channel bank for rx.
  txgain=3.0; This is to compensate for the drop in volume because of
; the -4 dB setting on the channel bank for tx.

  context=pstn
  rxgain=1.4; This was bumped up last, as a result of a milliwatt test.
;
  txgain=1.4; This was also bumped up, because it makes the outbound
; calls a bit louder, and doesn't seem to overdrive the 
; line.  I figure the gain loss on rx (which was calibrated
; with the milliwatt test) should be similar to tx gain lost,
; although I couldn't directly test this.

Now, when I turn on echo cancellation for all my interfaces, the echo is
completely gone.  After compensating for the gain drop on the channel banks
with asterisk boost, the call volumes sound good too.

-- 
Shaw Terwilliger [EMAIL PROTECTED]
SourceGear LLC


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[Asterisk-Users] ACD/queues question

[Pedro Nunes]

Hi there,



Does anyone know how to
setup an overflow queue? When a call rings on the queue A, if all agents were
busy, the call goes to the queue B.

If all agents in queue B were
busy, then the call stays on both queues until somebody answers it. 



I think this is a basic
ACD feature available on most PBX that support ACD functionality. 

Does anybody knows how to
do it with asterisk??





Thanks in advance





Pedro Nunes










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Re: [Asterisk-Users] MWI for endpoints not registered at Asterisk

[Ryan Hulsker]

Take a look at sipsak. http://sipsak.org/
Or at the wiki on how to use it.
http://www.voip-info.org/wiki-Asterisk+at+large

I think you will still need to be able to look up the IP address that
corresponds to your sip client though.

Ryan Hulsker


On Wed, 2005-10-12 at 08:21, Peter Bowyer wrote:
 On 12/10/05, Stojan Sljivic - Pamet [EMAIL PROTECTED] wrote:
  Hi,
 
  We have phones registered at another soft switch, and would like to use
  Asterisk as a Voicemail system.
  Is it possible and how to configure Asterisk to send NOTIFY messages (for
  MWI) to the endpoints that are not registered to the Asterisk?
 
 I couldn't find a way round this, and ended up using a 'spare' line
 presentation on my GXP-2000 phones to register to the voicemail server
 simply to pick up the NOTIFYs. Since the phone only has a single MWI
 LED, it doesn't matter which line the NOTIFY comes in on.
 
 Peter
 
 --
 Peter Bowyer
 Email: [EMAIL PROTECTED]
 Tel: +44 1296 768003
 VoIP: sip:[EMAIL PROTECTED]
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RE: [Asterisk-Users] RE: faxing to/from asterisk - new scripts

[Carlos Alperin]

Can you send me those scripts to calperinatsenecacom.net.?



Thanks in advance.



Carlos Alperin

Senior System Engineer

Seneca Communications, LLC

[EMAIL PROTECTED]











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Technical Support
Sent: Friday, October 07, 2005
10:55 AM
To: asterisk-users@lists.digium.com;
'Roman'
Subject: [Asterisk-Users] RE:
faxing to/from asterisk - new scripts





Roman: 









I created two bash scripts called Mail2Fax and Fax2Mail for
use with the asterisk sever.











They leverage the app_txfax and app_rxfax scripts, along
with ast_fax. They make using these apps a lot easier, including being
able to mail to [EMAIL PROTECTED] for outgoing
faxes and then extracting phone numbers from the subject line! (Makes it
easy to use with Sendmail without complex rules / virtual user tables).











They also include error logs, parameter checking, etc.











Let me know if you want them











Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your
business...









T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca 










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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Matthew T. O'Connor]

Ok, that would be helpful for me with some other problems, however I 
don't see keys anywhere in my sip.conf or my phoneXX.conf files.  I'm 
using the 1.5.2 Sip firmware the the conf files that came with that, so 
I don't have an ipmid.cfg file.  Is this something I can just add to my 
sip.conf?


Anyone out there any suggestions on how to do the speed dial in-call?

Thanks,

Matt



Mojo with Horan  Company, LLC wrote:
Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
itemfnPark/fnct#70#/ctsd3/sd/item) and then in 
ipmid.cfg:
keys key.IP_500.31.function.prim=SpeedDial 
key.IP_500.31.subPoint.prim=3/


This tells the phone to run Speed Dial 3 whenever the Services button 
(button #31 on a 500/501) is pressed.  I hope someone can help us 
configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:
I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional 
but it speeds it up.) to park a call.  Personally I think this is 
easy, but my users would like one button to do this for them.  The 
Polycom has buttons we don't need (Transfer  Services), it would be 
nice if I could remap one of those buttons to dial #70#.  Or if I 
could add a soft button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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Re: [Asterisk-Users] RE: faxing to/from asterisk - new scripts

[Thameem Ansari]

Please send them to my email [EMAIL PROTECTED]

Thanks,
ThameemOn 10/12/05, Carlos Alperin [EMAIL PROTECTED] wrote:
















Can you send me those scripts to calperinat
senecacom.net.?



Thanks in advance.



Carlos Alperin

Senior System Engineer

Seneca Communications, LLC

[EMAIL PROTECTED]












From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Technical Support
Sent: Friday, October 07, 2005
10:55 AM
To: asterisk-users@lists.digium.com;
'Roman'
Subject: [Asterisk-Users] RE:
faxing to/from asterisk - new scripts





Roman: 









I created two bash scripts called Mail2Fax and Fax2Mail for
use with the asterisk sever.











They leverage the app_txfax and app_rxfax scripts, along
with ast_fax. They make using these apps a lot easier, including being
able to mail to [EMAIL PROTECTED] for outgoing
faxes and then extracting phone numbers from the subject line! (Makes it
easy to use with Sendmail without complex rules / virtual user tables).











They also include error logs, parameter checking, etc.











Let me know if you want them











Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your
business...









T: (519) 672-8238
E: 
[EMAIL PROTECTED]
W: 
www.ocg.ca
 











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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Mojo with Horan Company, LLC]

Do you already have an ipmid/ipmid block in your sip.cfg?  add the 
keys ... / in there:

Try putting:
ipmid
  ...
  ...
  keys key.IP_500.31.function.prim=SpeedDial 
key.IP_500.31.subPoint.prim=3/

/ipmid

Moj

Matthew T. O'Connor wrote:
Ok, that would be helpful for me with some other problems, however I 
don't see keys anywhere in my sip.conf or my phoneXX.conf files.  I'm 
using the 1.5.2 Sip firmware the the conf files that came with that, so 
I don't have an ipmid.cfg file.  Is this something I can just add to my 
sip.conf?


Anyone out there any suggestions on how to do the speed dial in-call?

Thanks,

Matt



Mojo with Horan  Company, LLC wrote:

Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
itemfnPark/fnct#70#/ctsd3/sd/item) and then in 
ipmid.cfg:
keys key.IP_500.31.function.prim=SpeedDial 
key.IP_500.31.subPoint.prim=3/


This tells the phone to run Speed Dial 3 whenever the Services button 
(button #31 on a 500/501) is pressed.  I hope someone can help us 
configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:

I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional 
but it speeds it up.) to park a call.  Personally I think this is 
easy, but my users would like one button to do this for them.  The 
Polycom has buttons we don't need (Transfer  Services), it would be 
nice if I could remap one of those buttons to dial #70#.  Or if I 
could add a soft button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] detect SIP phone availability before dialing

[Mojo with Horan Company, LLC]

This is a pretty popular question.  IIRC SIP phones can't tell you their 
statuses, you need to send a call to them and determine whether or not 
they're Busy Now...


[EMAIL PROTECTED] wrote:

Hello,

 I need to detect availability of SIP phone before dialing. I need to
know if phone is 
 BUSY, CHANUNAVAIL before dialing. If phone is free, then I will dial

it.

 I need for automatic callback (.call files), but I need to know if it
is available both
 SIP phones before calling.

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(907) 747- x112
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[Asterisk-Users] displaying a message on the Snom 320 using sipsak

[Franklin Webb]

Greetings fellow list members,
 It seems like a lot of 
peoplehave been having trouble getting indicators workingon the Snom 
phones, myself included. Recently I was able to get the "desktop" 
functionality of sipsak to work on my Snom320, and I thought I would share what 
I could with the list. For those not familiar this will replace the 
standard display when you are not on a call (normally showing the registered 
extension) with a text message of your choosing. Our intent is to update 
this when our agents log into, and out of, queues. This will give a visual 
indicator for agents and supervisors in our call center as to whether or not the 
phone is logged in, which is a large concern for us, and probably any call 
center.

For the record I tried this with a Snom360 also and 
could not get it working.

1. Setup the phone in Asterisk as 
normal
2. Get and install sipsak. It can be 
found at http://sipsak.org/(can be on any 
machine on your network, we used a Fedora Core 3 machine for this).
3.In the Snom320 Configuration, under 
the"SIP" tab of your extensions line (Line 1 for me) make sure "Support 
Broken Registrar" is set to"on"
4. In the Snom320 Configuration, 
under"Advanced" make sure "Filter Packets from Registrar" isset to 
"off"
5. In the Snom320 Configuration, under 
"Advanced"under "Networkidentity (port):" set it to "5060" (you 
might be able to usea different port in here and in the sipsak command, 
butthis is what worked for me.
6. Reboot the phone (just to be sure the settings 
take)

Then use the following sipsak command:

sipsak -vvv -M -O desktop -B "Test Msg" -r 5060 -s 
sip:[EMAIL PROTECTED]

where:
 "Test Msg" is the message you 
want displayed. To turn the message off just set it to empty string 
("").
 5060 is the port, you could try 
another port here if you set your phone to another port under 
"Advanced"
 6670 is the extension of the 
phone
 192.168.51.251 is the IP of the 
PHONE, not the Asterisk server. It does not appear that you can use the IP 
of the Asterisk server.

You can get a list of phones with IPs using the 
Asterisk command "sip show peers". Our intent is to build a simple 
database matching extension to IP and then execute sipsak commands from a 
script, probablyin the manager API, when agents log in and out that will 
update the phonedisplay accordingly.

I hope this is helpful to some of you.

Franklin Webb
InterMedia Marketing Solutions

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Re: [Asterisk-Users] E400P vs te410p vs te411p

[MvPhone]

Hi,

Check out http://store.pbxhardware.com = it has better prices on the
E400P / T400P cards. There are also 2 port versions of these.

The difference between the TE4XX cards is there is no echo canceller
and the PCI chipset doesn't handle the master mode - that eats a
little bit of CPU time.

regards
Martin

Hi,

I found E400P quad PRI card quite cheap (749USD):

http://www.govarion.com/product_info.php?cPath=1products_id=2osCsid=68cdd6e3d08754

in comparison to te410p (approx 1500 USD )

http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P

Now newer generation with HW echo canceling emerged (te411p).

I'm not sure in what things those two cards differ and what would be
best
option to buy (I believe there is big performance gap between them, but
don't know how big and if it's worth of money) Also how do you find
HW
echo canceling in te411p ?

Any advice, help ?

Thanks in advance,

regards,

Rob.
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[Asterisk-Users] Bulk Buys/Group Buys

[Nathan Pralle]

Hey folks,

Anyone know of companies selling bulk SIP adaptors (phones, adaptors, 
etc.) or has the list ever considered doing something like a bulk buy?


I was just curious...I'm looking to get another 5-6 Grandstreams or 
similar and I figured I'd ask the list.  If we found something that lots 
of people wanted, it probably couldn't hurt to contact a company and ask 
for bulk deals.


Whadya think?  Anyone tried this before?

Nathan

--
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[Asterisk-Users] SIP behind NAT to pub Asterisk, best solution?

[Blake Krone]

What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices.


What are people using? STUN? SER?

Thanks in advance!

-blake
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Re: [Asterisk-Users] Dial DTMF after bridging call

[Dinesh Nair]

On 10/12/05 15:41 Corey Frang said the following:
Interestingly, I started playing with the numbers on my phone after the 
dial messed up, and I could get the DTMF tones stuck playing one tone 
for a long time.  If i took the D() out of it It didn't have that problem.


On Aug 25, 2005, at 15:04, Joseph wrote:


Is there a way to dial DTMF after bridging the call.
The current option D() in Dial will dial DTMF before the call is bridged
and this doesn't do the job.
I need to dial DTMF after the call is bridged and the message is played
with Background


*CLI show application senddtmf

  -= Info about application 'SendDTMF' =-

[Synopsis]
Sends arbitrary DTMF digits

[Description]
  SendDTMF(digits[|timeout_ms]): Sends DTMF digits on a channel.
  Accepted digits: 0-9, *#abcd
 Returns 0 on success or -1 on a hangup.

--
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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Matthew T. O'Connor]

While I don't have it working yet, I think I have it figured out.  I 
have to add keys / entries to my sip.conf  Based on your example I was 
able to find the relevant info in the Polycom SIP 1.5 Admin Guide 
section 4.6.1.15.


My next question, which I haven't found in the admin guide (at least not 
yet) is where to you get a list of the buttons and their respective numbers?


Thanks again,

Matthew


Mojo with Horan  Company, LLC wrote:
Do you already have an ipmid/ipmid block in your sip.cfg?  add the 
keys ... / in there:

Try putting:
ipmid
  ...
  ...
  keys key.IP_500.31.function.prim=SpeedDial 
key.IP_500.31.subPoint.prim=3/

/ipmid

Moj

Matthew T. O'Connor wrote:
Ok, that would be helpful for me with some other problems, however I 
don't see keys anywhere in my sip.conf or my phoneXX.conf files.  
I'm using the 1.5.2 Sip firmware the the conf files that came with 
that, so I don't have an ipmid.cfg file.  Is this something I can 
just add to my sip.conf?


Anyone out there any suggestions on how to do the speed dial in-call?

Thanks,

Matt



Mojo with Horan  Company, LLC wrote:

Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
itemfnPark/fnct#70#/ctsd3/sd/item) and then in 
ipmid.cfg:
keys key.IP_500.31.function.prim=SpeedDial 
key.IP_500.31.subPoint.prim=3/


This tells the phone to run Speed Dial 3 whenever the Services 
button (button #31 on a 500/501) is pressed.  I hope someone can 
help us configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:

I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional 
but it speeds it up.) to park a call.  Personally I think this is 
easy, but my users would like one button to do this for them.  The 
Polycom has buttons we don't need (Transfer  Services), it would 
be nice if I could remap one of those buttons to dial #70#.  Or if 
I could add a soft button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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[Asterisk-Users] send Q931 information element keypadfacility ?!

[Bruno Voigt]

Hi all,

I'm looking for a way with any asterisk-version with TE410P (cpe
EuroISDN, Q931)
for sending an INFORMATION ELEMENT KeypadFacility,
eg. *87, during a connected call to the PSTN switch.

Are there existing functions in asterisk to generate  send such IE ?

If not what existing modules would be best to derive from?

TIA,
Bruno

begin:vcard
fn:Bruno Voigt
n:Voigt;Bruno
org:IC3S AG
adr:;;Baeckerbarg 6;Wilstedt;;D-22889;Germandy
email;internet:[EMAIL PROTECTED]
tel;work:+494109555105
tel;fax:+4941095
tel;cell:+4970068600686
x-mozilla-html:FALSE
url:http://www.ic3s.de
version:2.1
end:vcard

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Re: [Asterisk-Users] Bulk Buys/Group Buys

[Ariel Batista]

Nathan Pralle wrote:

Hey folks,

Anyone know of companies selling bulk SIP adaptors (phones, adaptors,
etc.) or has the list ever considered doing something like a bulk buy?


Give a call to VoipSupply.com 800-398-VOIP (8647)


I was just curious...I'm looking to get another 5-6 Grandstreams or
similar and I figured I'd ask the list.  If we found something that
lots of people wanted, it probably couldn't hurt to contact a company
and ask for bulk deals.

Whadya think?  Anyone tried this before?

Nathan

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[Asterisk-Users] Feature codes work on SIP phone but not analog?

[Doug]

Hi,

This is what I have in extensions_custom.conf:

; Time of Day functionality:
exten = *60,1,Answer
exten = *60,2,Wait(1)
exten = *60,3,SayUnixTime(,,IMSP)
exten = *60,4,Hangup

It works on a Cisco 7940 IP Phone, but on
analog phones, when I dial *60, I just
get a dial tone.  If I dial *60#, then
I just get a fast busy.

What's going on?

Also, how can I get Feature Codes to work
for all contexts?

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Re: [Asterisk-Users] SIP behind NAT to pub Asterisk, best solution?

[chentschel]

Mensaje citado por: Blake Krone [EMAIL PROTECTED]:

 What is the best solution? I dont want to have modify firewall\'s at all or
 do port fowarding. Ideally I would like a solution that with either a
 softphone or wireless hardphone one could connect via friends, family, or
 hotspots without reconfiguring their devices.
  What are people using? STUN? SER?
  Thanks in advance!
  -blake
 
Give a try to the sip-helper for netfilter, and please let me know if this 
works for ya. 
Thanks. 
Christian. 
__
Registrate desde 
http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y 
participá de todos los beneficios del Portal Arnet.
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Re: [Asterisk-Users] detect SIP phone availability before dialing

[Paul Zimm]

Use application ChanIsAvail with the s option. This option only exists 
in CVS-HEAD version, the 1.0.x versions don't have this option.


from documentation:

If the option 's' is specified (state), will consider channel unavailable
when the channel is in use at all, even if it can take another call.


This is a pretty popular question.  IIRC SIP phones can't tell you 
their statuses, you need to send a call to them and determine whether 
or not they're Busy Now...


[EMAIL PROTECTED] wrote:


Hello,

 I need to detect availability of SIP phone before dialing. I need to
know if phone is  BUSY, CHANUNAVAIL before dialing. If phone is 
free, then I will dial

it.




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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Mojo with Horan Company, LLC]

They are in the 1.5 admin guide, pages 22-25

Matthew T. O'Connor wrote:
While I don't have it working yet, I think I have it figured out.  I 
have to add keys / entries to my sip.conf  Based on your example I was 
able to find the relevant info in the Polycom SIP 1.5 Admin Guide 
section 4.6.1.15.


My next question, which I haven't found in the admin guide (at least not 
yet) is where to you get a list of the buttons and their respective numbers?


Thanks again,

Matthew


Mojo with Horan  Company, LLC wrote:

Do you already have an ipmid/ipmid block in your sip.cfg?  add the 
keys ... / in there:

Try putting:
ipmid
 ...
 ...
 keys key.IP_500.31.function.prim=SpeedDial 
key.IP_500.31.subPoint.prim=3/

/ipmid

Moj

Matthew T. O'Connor wrote:

Ok, that would be helpful for me with some other problems, however I 
don't see keys anywhere in my sip.conf or my phoneXX.conf files.  
I'm using the 1.5.2 Sip firmware the the conf files that came with 
that, so I don't have an ipmid.cfg file.  Is this something I can 
just add to my sip.conf?


Anyone out there any suggestions on how to do the speed dial in-call?

Thanks,

Matt



Mojo with Horan  Company, LLC wrote:


Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
itemfnPark/fnct#70#/ctsd3/sd/item) and then in 
ipmid.cfg:
keys key.IP_500.31.function.prim=SpeedDial 
key.IP_500.31.subPoint.prim=3/


This tells the phone to run Speed Dial 3 whenever the Services 
button (button #31 on a 500/501) is pressed.  I hope someone can 
help us configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:


I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional 
but it speeds it up.) to park a call.  Personally I think this is 
easy, but my users would like one button to do this for them.  The 
Polycom has buttons we don't need (Transfer  Services), it would 
be nice if I could remap one of those buttons to dial #70#.  Or if 
I could add a soft button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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[Asterisk-Users] PPP over ISDN PRI usinf Asterisk

[gshaw]

Hi
Is anyone using Asterisk for PPP over PRI ISDN. Any example would be
appreciated. I saw ZAPRAS and PPPD commands. The documentation has
zaptel.conf example for PPP using T1/E1 clear channel.

Regards
Goutam



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RE: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Andy Goss]

It is on page 22 and 23 in my admin guide.

Andy

--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor
 Sent: Wednesday, October 12, 2005 2:38 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Polycom: Button Remapping, HELP!
 
 While I don't have it working yet, I think I have it figured out.  I
 have to add keys / entries to my sip.conf  Based on your example I
was
 able to find the relevant info in the Polycom SIP 1.5 Admin Guide
 section 4.6.1.15.
 
 My next question, which I haven't found in the admin guide (at least
not
 yet) is where to you get a list of the buttons and their respective
 numbers?
 
 Thanks again,
 
 Matthew
 
 
 Mojo with Horan  Company, LLC wrote:
  Do you already have an ipmid/ipmid block in your sip.cfg?  add
the
  keys ... / in there:
  Try putting:
  ipmid
...
...
keys key.IP_500.31.function.prim=SpeedDial
  key.IP_500.31.subPoint.prim=3/
  /ipmid
 
  Moj
 
  Matthew T. O'Connor wrote:
  Ok, that would be helpful for me with some other problems, however
I
  don't see keys anywhere in my sip.conf or my phoneXX.conf files.
  I'm using the 1.5.2 Sip firmware the the conf files that came with
  that, so I don't have an ipmid.cfg file.  Is this something I can
  just add to my sip.conf?
 
  Anyone out there any suggestions on how to do the speed dial
in-call?
 
  Thanks,
 
  Matt
 
 
 
  Mojo with Horan  Company, LLC wrote:
 
  Matthew, when I tried this, I couldn't get the soundpoints to dial
  in-call.  They thought there were picking up a new line for a new
 call.
 
  I created a speed-dial entry (in MACADDRESS-directory.xml,
  itemfnPark/fnct#70#/ctsd3/sd/item) and then in
  ipmid.cfg:
  keys key.IP_500.31.function.prim=SpeedDial
  key.IP_500.31.subPoint.prim=3/
 
  This tells the phone to run Speed Dial 3 whenever the Services
  button (button #31 on a 500/501) is pressed.  I hope someone can
  help us configure them now to dial these digits in-call...
 
  Mojo
 
  Matthew T. O'Connor wrote:
 
  I need to find a way to have the Polycom phones automatically
park
  calls.  Right now my users hit #70# (I know the last # is
optional
  but it speeds it up.) to park a call.  Personally I think this is
  easy, but my users would like one button to do this for them.
The
  Polycom has buttons we don't need (Transfer  Services), it would
  be nice if I could remap one of those buttons to dial #70#.  Or
if
  I could add a soft button during a call that would work too.
 
  Anyone have any suggestions on how to do this?
 
  Thanks much,
 
  Matthew O'Connor
 
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[Asterisk-Users] Is it possible to listen and respond on more than one IAX port?

[Steve Gladden]

Hello,

I'd like to know if it is possible to get * to listen and respond on more
than just one single udp port.

I've run into several situations where I'd like IAX to work on an alternate
port as well as be able to work on the standard port.

I'm wondering if there is a way to do this?

Thanks!!

Steve


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[Asterisk-Users] Canadian Association of VoIP Providers

[John Lange]

My apologies for the cross-posting.

If you are a business or individual providing Voice over IP services in
Canada then we encourage you to read this email carefully otherwise
please disregard.

-

As you are most likely aware, the CRTC has undertaken the roll of
regulating VoIP services in Canada and is currently conducting hearings
with the goal of putting in place regulatory requirements for all VoIP
providers.

Specifically, the CRTC's CISC VoIP 911 working group
( http://www.crtc.gc.ca/cisc/eng/cisf3e4_20.htm ) is very actively
looking at what regulations to put in place in order to implement E911
services for VoIP.

The recommendations of this committee will have direct impact on your
business. Currently this working group is is largely comprised by the
Local Exchange Carriers (ILECs  CLECs) with representation from the
large VoIP providers (Primus  Vonage). To date only a very few smaller
VoIP providers are participating.

Subsequently, much of the discussion is oriented around solutions
designed to work in the traditional telco world. Depending on your
companies infrastructure these solutions may be very expensive or
completely impossible for your business to implement.

Some members of the working group are even of the position that VoIP
service be abolished altogether.

Your companies direct participation in the hearings is the best way to
have an impact. However, we acknowledge that not all companies have the
time and/or resources to fully participate lengthy public hearings.

It is with this in mind we propose the formation of a Canadian industry
association for VoIP providers and we invite you to participate.

The short term goal is to contact and organize Canadian VoIP providers
into a formal association.

Longer term the association will work towards the following goals:

- Keep VoIP providers informed about current regulatory issues
- Ensuring VoIP providers have a place at the CRTC table
- Develop industry recommendations
- Communicate industry recommendations to the CRTC working group
- Communicate industry positions to the media
- Other (to be determined by the association)

At the outset it is envisioned that this group would work in the
following way:

- No membership fee
- Regular updates via email list
- Frequent Conference calls
- No face-to-face meetings (no travel)
- Development of an Industry web site
- In-person representation at each CRTC meeting (The CRTC working group
meets monthly in a different province each month. We hope to have at
least one member representative attend each meeting.)

To voice your support (or opposition) for the formation of this group
please contact me directly either by email or telephone (contact
information in the signature).

It is important that you do not delay. CISC working group
recommendations to the CRTC are forthcoming.

You will be contacted with details on how to participate in the
formation of this association. Our intention is to hold our first
conference call as early as possible (early next week).

NOTE: No web site or association material yet exists because the group
has not been officially formed and named. This will be one of the first
items of business for the new group.

Regards,
-- 
John Lange
President OpenIT ltd. www.Open-IT.ca (204) 885 0872
VoIP, Web services, Linux Consulting, Server Co-Location

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[Asterisk-Users] RE: faxing to/from asterisk - new

[Technical Support]

The fax2mail and 
mail2fax scripts can be found on www.generationd.com



Michelle 
DupuisTechnical Support SpecialistOxford Consulting Group Ltd.Making IT work for your 
business...

T: (519) 672-8238E: 
[EMAIL PROTECTED]W: 
www.ocg.ca 

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Re: [Asterisk-Users] parameters documentation

[FELIX E SKOWRONEK]

The people who have been documenting Asterisk have been working on a book 
for the last few months, it has been published by O'reilly (Asterisk-The 
Future of Telephony)and is just now finding it's way into the major 
bookstores, listed under Open-Source at BarnsNoble.


While it will not answer everything asterisk can do, but its glossery and 
and appendix are very helpful for quick reference.  If you have been 
following the Asterisk Documentation Project, some of it will be old hat, 
but I'm looking forward to replacing my huge stack of printouts with it.


It gives a pretty good overview of VOIP, Networking, Telephony, etc.

http://www.oreilly.com/catalog/asterisk/



From: Steve Totaro [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [Asterisk-Users] parameters documentation
Date: Wed, 12 Oct 2005 11:17:01 -0400

There is plenty of documentation online for both the 3com and *.  You have
to have good search skills I guess.

3com has the best knowledge base I have seen.
http://knowledgebase.3com.com/ and there are tons of 3com dealers that can
help.

I think you may need to learn some basic networking before learning
asterisk.  NAT is a very basic concept in networking as well as ports such
as 5060 (standard port for SIP).

There is a very steep learning curve for asterisk and networking in 
general.

If you want to learn it then you need to dig into the wiki and read all the
posts that come across the user's list (well maybe not all  of them).

There are plenty of consultants that you can hire if you are not up to it.


- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; 
[EMAIL PROTECTED]

Sent: Wednesday, October 12, 2005 10:34 AM
Subject: Re: [Asterisk-Users] parameters documentation


 I come from a NBX100
 No documentation available.
 1 day it starts saying: syslog full and voicemail stop working
 No one was able to tell me what was the meaning of that alert
 .
 3COM NBX anyway is a good product, but the price is too high, especially 
4

 years ago, and especially the price of the telephone is very high.

 Andrea





  asterisk
  [EMAIL PROTECTED]
  echnologies.com   
To

  Sent by:  Asterisk Users Mailing List -
  asterisk-users-bo Non-Commercial Discussion
  [EMAIL PROTECTED] asterisk-users@lists.digium.com
  m.com  
cc



Subject

  13/10/2005 16.13  Re: [Asterisk-Users] parameters
documentation

  Please respond to
   Asterisk Users
   Mailing List -
   Non-Commercial
 Discussion
  [EMAIL PROTECTED]
  ists.digium.com






  I really hope this project will be implemented, without documentation
 evrything is too hard

 Not for the thousands of people that have figured it out.

 3Com NBX might be more your speed and plenty of documentation.



  Really strange answer. I am non used to search on playboy.com.
 
  Anyway, if you try to search
  insecure=very
  on www.voip-info.org, you find 742 links , a bit more for me. (I just
 want
  to know what it means)
 
  Moreovere, the first 20 links are non accessible at all
 
 

http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecurediff=6

 
  they speak about tiki-pagehistory.php, which appears not to exist.
 
  no other comments about this.
  
 
  I know about one project , asterisk documentation project
 
  http://www.asteriskdocs.org
 
  in its home page, the first line is
 
 
 
 
 
   Great software needs great documentation.
 
 
 
 
  I really hope this project will be implemented, without documentation
  evrything is too hard
 
  Andrea
 
 
 
 
   Steve Totaro
   [EMAIL PROTECTED]
   echnologies.com
 To
   Sent by:  Asterisk Users Mailing List -
   asterisk-users-bo Non-Commercial Discussion
   [EMAIL PROTECTED] 
asterisk-users@lists.digium.com

   m.com
 cc
 
 
 Subject
   12/10/2005 14.53  Re: [Asterisk-Users] parameters
 documentation
 
   Please respond to
Asterisk Users
Mailing List -
Non-Commercial
  Discussion
   [EMAIL PROTECTED]
 

Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Bob Goddard]

On Wednesday 12 Oct 2005 14:53, Tom Rymes wrote:
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Bob Goddard
  Sent: Tuesday, October 11, 2005 6:35 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
  are fax-capable?
 
  On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
   Tom Rymes wrote:
Frankly, I would recommend that you forget about trying
 
  to fax with
 
Asterisk. Buy a good Multitech analog modem and install HylaFAX.
   
Use the right tool for the job!!!
  
   Actually, you can use HylaFAX and Asterisk together.
  
 https://sourceforge.net/projects/iaxmodem/
  
   Just be certain that your audio path doesn't run over any
 
  lossy medium
 
   (so run IAXmodem on your Asterisk box).
 
  I'll expand on what Tom meant
 
  Use a hardware based DSP for faxing not software based.

 Actually Bob, that isn't what I meant. Lee simply suggested a different
 way (IAXModem instead of analog modem) of implementing what I meant. I
 would still recommend using analog if you can but, if you cannot, use
 IAXModem from Lee.

Okay!

 Asterisk's faxing capabilities are not nearly as advanced, stable, or
 easy to set up as HylaFAX. Also, there seem to be many problems with
 frame slipping and the like that screw up faxing over Digium cards, and
 maybe others as well.

Does Hylafax do software based faxing? As far as I knew, it has always
required a DSP.

 Either way, I was just saying that grabbing a good modem (see HylaFAX
 list archives for suggestions - NOT USRobotics!!!) and installing
 HylaFAX would be easier, more reliable, and all-in-all, a better
 solution than messing with Asterisk's built-in fax capability.

In other words, don't use a soft fax.
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