[Asterisk-Users] I can't resume a call on hold from zap device

[Marco Maiolini]

I have a strange problem: if I put on hold an incoming call from my Digium 
TE110P, I can't resume it and the person at the phone continues to hear MOH 
until the line falls.
My TE110P is connected with an italian E1 NT.
If I put on hold a call on a SIP channel I can resume it without any problems.

Is there someone that can help me?

These are my configurations:

zaptel.conf

span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
loadzone = it
defaultzone=it


zapata.conf:

[trunkgroups]

[channels]

language=it
signalling=pri_cpe
switchtype=euroisdn
usecallingpres=yes
pridialplan=local
prilocaldialplan=local
nationalprefix=0
internationalprefix=00
faxdetect=both
callwaiting=yes
echocancel=yes
immediate=no
overlapdial=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
group=1
context=from-pstn
channel = 1-15,17-31


Thanks in advance, Marco.


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Re: [Asterisk-Users] priorityjumping=no

[Olle E Johansson]

14 mar 2006 kl. 01.45 skrev Steve Kennedy:


On Mon, Mar 13, 2006 at 07:38:01PM -0500, Watkins, Bradley wrote:


That depends on what you mean by default.  The supplied sample
extensions.conf contains the priorityjumping=no by default, but if  
this

parameter is absent then the default is to jump n+101.


OK, that explains it, just wondering why the sample extensions.conf
turns it off, while the O'Reilly Asterisk book and alomst  
everything you

see on the web uses it ???

I would have thought the default would be to have it on?


In 1.2 we switched from priority jumping to returning values in
STATUS variables. If you look in the sample extensions.conf you
will see how that work, that the result of dial() is now returned in
the DIALSTATUS variable.

Using DIALSTATUS will give you many more ways to control the
result of dial, than just checking for +101. You can react to no answer,
congestion, busy, privacy and other things.

Even more detailed status is frequently available in the HANGUPCAUSE
variable that uses ISDN cause codes.

Show application dial is your best friend.

For each application that used to return priority+101 or something else
there's a new STATUS variable that you can use.

We do no longer accept patches that use priority jumping, that's
a remain from the past that will be removed in future versions of
Asterisk. From the file UPGRADE.txt in the svn trunk distribution:

Applications:

* In previous Asterisk releases, many applications would jump to  
priority n+101
  to indicate some kind of status or error condition.  This  
functionality was
  marked deprecated in Asterisk 1.2.  An option to disable it was  
provided with
  the default value set to 'on'.  The default value for the global  
priority

  jumping option is now 'off'.

/Olle



---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/



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Re: [Asterisk-Users] Analog Desktop Phone

[fischer]

Dan,

what is so wrong with the snom360 ? I now your wiki website, but as far as I 
can see, nearly all major issues are resolved. Meanwhile we have the version 
5 branch much more stabilized, see beta 5.5:

http://www.snom.com/wiki/index.php/Beta_Firmware

and if you don't like to use a beta, with release version 4.5 you can work 
reliably.

Best regards,

Sven

On Monday 13 March 2006 19:38, [EMAIL PROTECTED] wrote:
 On Mon, 13 Mar 2006, Kerry Garrison wrote:
  system) you do NOT want to use a cheap phone on this system. At a minimum
  go with a Linksys SPA941 or a Snom 360. You will have either one working
  in a

 I would wait until snom fixes the issues with the 360 firmware.

 -Dan
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-- 
---
See our FAQs at: http://www.snom.com/faq0.html?L=1
Whitepapers at:  http://www.snom.com/white_papers.html
---
snom technology AG   Gradestraße 46 D-12347 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED]   http://www.snom.com
---
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Re: [Asterisk-Users] Asterisk RealTime Question, Please help

[Peter Spikings]

Hi,

OK, that will enable the auto generation of a context but as the new
context won't have a switch statement it doesn't help with this
problem... I may try writing a default switch if no matching context
found type patch.

Peter.

On Mon, 2006-03-13 at 20:51 +0200, Benchev wrote:
 I was able to install Asterisk and Asterisk-addons and use them
   successfully. But I have a problem now, I have many contexts and it looks
   like Asterisk is unable to find the context given directly in Mysql DB
   unless I specify it in Extensions.conf to switch it to RealTime. If I add
   a new context in Mysql then I have to add it in Extensions.conf and
   reload extensions whenever I need a new context. Please tell me if there
   is a way to avoid all this and make Asterisk take contexts directly from
   Mysql without mentioning that context in Extensions.conf. If this is
   possible then I can make my Asterisk RealTime actually and modify
   contexts directly in Mysql.
 Idea from the wiki:
 ; If regcontext is specified, Asterisk will dynamically create and destroy a
 ; NoOp priority 1 extension for a given peer who registers or unregisters with
 ; us.  The actual extension is the 'regexten' parameter of the registering
 ; peer or its name if 'regexten' is not provided.  More than one regexten may
 ; be supplied if they are separated by ''.  Patterns may be used in regexten.
 ;
 ;regcontext=sipregistrations
 That means that you should creat a mother context in extensions.conf:
 [sipregistrations]
 
 But first I would try to add a field regcontext along with regexten(which 
 already there) in sip_users table since for the trick to work ...
 read http://www.voip-info.org/wiki-Asterisk+sip+regcontext
 
 Hope this will give you a clue.
 Benchev
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[Asterisk-Users] Inbound sipgate number forwarding to differnet users

[Francois-Xavier Bas]

How can I forward my offcial sipgate number to different users, I would
like to know if it is possible to append  a local user number to my
official number when dialing, then in this way it could be forwarded
using the suffixe local user number.The prefixe number would be the
official sipgate number.Are tehre some companies that provide this kind
of service instead of suscribing to many official numbers?

-- 
Francois-Xavier Bas

RSS-Global Technologies Ltd.
Bachemer Strasse 266
50935 Cologne
Germany

phone:  +49221 297-6491
email:  [EMAIL PROTECTED]
url:www.rss-global.com



begin:vcard
fn:Francois Bas
n:Bas;Francois
org:RSS Global Technologies Ltd.
adr:;;Bachemer Strasse 266;Cologne;;50935;Germany
email;internet:[EMAIL PROTECTED]
title:System Administrator
tel;work:+49 221 2976 491
x-mozilla-html:TRUE
url:http://www.rss-global.com
version:2.1
end:vcard

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[Asterisk-Users] DATA CALLS annoying my system

[Pisac]

When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise
type of call, but answering anyway (playing IVR messages, ringing
phones, etc...)

How to stop that? I want that only VOICE calls are answered, and
DATA/FAX to be ignored.

(I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f,  ZapHFC)

Log:

-- Accepting data call from '' to '3001' on channel 0/2, span 1
-- Executing Answer(Zap/2-1, ) in new stack
-- Executing BackGround(Zap/2-1, ivr_intro) in new stack


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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

[Chris Stenton]

Yes it does display caller id as callingnumber@ip of calling party but 
that does not interfere with me hitting dial from missed calls. Seems the 
Cisco phone sends the  sip INVITE as  callingnumber@ip of calling party 
rather than callingnumber@ip address of defaultproxyserver but asterisk 
ignores the info after the @?


Chris

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

Sent: Monday, March 13, 2006 11:45 PM
Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time



The 8.2 firmware displays Caller ID as callingnumber@proxyaddr...
this becomes problematic for users that want to dial from their
'Missed Calls' log.

Omar

On 3/13/06, Nathan Bowyer [EMAIL PROTECTED] wrote:

On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote:
 I have had no issues with 8.2 so far!

 Chris


Except the Caller ID issue reported in another thread?

 
  This issue has been fixed in SIP firmware 7.5
 
  Omar A. Sabek
 
  Yes, and I read that SIP 7.5 firmware have some other issues. They
  recommend using 7.4 firmware. I'm not sure how good in new 8.2 
  firmware.

 
 
  Tomislav
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Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?

[Chris Stenton]

Maybe I have something strange in my dial plan but I have no problem just 
hitting dial from missed calls under 8.2.


Chris

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

Sent: Monday, March 13, 2006 8:44 PM
Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?


We rolled back to 7.4 cause of that too.  7.5 has a strange bug where if 
the server loses connection, the phone's just don't try re-registering.


Aaron

Tim Connolly wrote:

Just curious, why not 7.5 ? -Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Monday, March 13, 2006 2:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?


I'm using P0S3-08-2-00.. I noticed the callerID started showing

up
with the number, then @proxy-addr... So the callerID on the phone 
looks like: [EMAIL PROTECTED] which of course is logged in the 
missed calls exactly like that, and completely foobars the dialing 
string if you try to dial a missed call by simply hitting the dial 
button. Can anyone else verify this problem?


Yeah, that bothered me so I rolled back to SIP 7.4.

Nabeel

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Re: [Asterisk-Users] music on hold without mpg123

[Chris Stenton]

I have found in the past that using the  resample -ql option gives better 
results.


Chris

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

Sent: Tuesday, March 14, 2006 7:12 AM
Subject: Re: [Asterisk-Users] music on hold without mpg123




Hi Matt,
thank you for your inputs. I updated the page at 
http://www.oinko.net/astrecipes/index.php?n=152 so that it merges in the 
new pointers you sent me.


I also think that converting to the native codec format will be the best 
choice - you can usually trade in some disk space for performance. In any 
case, even having everything as WAV will require only a relatively cheap 
encoding, and not the full mp3 - slin - {your codec} encoding plus the 
external process that mpg123 used to require.


I wonder: is there any way to use Asterisk as a transcoding tool, i.e. 
using its internal transcoding capabilities to transcode a given file to 
all its supported codecs? (I'm thinking of g729 and ilbc, things that are 
not supported by Sox but that might be useful in a real-life scenario)


Thank you
l.

In data Mon, 13 Mar 2006 19:29:57 +0100, Matt Roth [EMAIL PROTECTED] ha 
scritto:


Lenz,

This method is referred to as file-based or native MOH, and I have some 
additional information regarding it.  First, a short post on why we 
moved from the rawplayer method to native MOH on our production box, 
with a quote from Kevin Fleming regarding the impact the change would 
have on scalability.


 - 
http://lists.digium.com/pipermail/asterisk-users/2006-February/141180.html


Second, I *believe* (please correct me if I'm wrong) that in order to 
get the full benefits of native MOH, the music files should be converted 
to the codec that the calls will be in.  This allows Asterisk to play 
MOH without performing any transcoding, which lowers the resource 
utilization on the box.  Here is a guide for converting WAV files to the 
desired codec, which addresses the four characteristics that describe 
audio data.


- http://lists.digium.com/pipermail/asterisk-users/2006-March/142108.html

Please feel free to add any of this information to your site and don't 
hesitate to contact me if you spot any mistakes.


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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--
Assum est, versa et manduca.
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Re: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x

[Jeroen Zwarts]

Reply to self:
Last week I had some time to figure out a workaround for the CDR logging
problem. I used an AGI-script together with de Mysql CLI application. It is
far from perfect, and I want to spend some more time to figure out a better
way, but this seems to be working OK on my testmachine:

This is what I do:

I run a DeadAGI script in the Hangup extension like this:

---
exten = h,1,Deadagi (fixcdr.sh)
exten = h,2,Hangup()
---

The /var/lib/asterisk/agi-bin/fixcdr.sh looks like this:

--
#!/bin/bash

read agi_request
read agi_channel
read agi_language
read agi_type
read agi_uniqueid
read agi_callerid
read agi_calleridname
read agi_callingpres
read agi_callingani2
read agi_callington
read agi_callingtns
read agi_dnid
read agi_rdnsid
read agi_context
read agi_extension
read agi_priority
read agi_enhanced
read agi_accountcode

set -- $agi_uniqueid
uniqueid='$2'

set -- $agi_callerid
telefoonnummer='$2'

echo -e  mysql --user=asterisk --password=asterisk --exec='use cdr;UPDATE
cdr SET src=\042$telefoonnummer\042 WHERE uniqueid=\042$uniqueid\042'   
/test/runthis.sh
echo -e  mysql --user=asterisk --password=asterisk --exec='use cdr;UPDATE
cdr SET clid=\042$telefoonnummer\042 WHERE uniqueid=\042$uniqueid\042'   
/test/runthis.sh
--

And then I let cron run the runthis.sh script. And the runthis.sh executes
the SQL strings in the file with the mysql monitor application.
The place where you put the runthis.sh must be writeable for the user that
Asterisk runs on (in my situation running Gentoo it is the user asterisk
with group asterisk)

As I already told, it is far from perfect (I have to find a way to empty the
runthis.sh file after cron has taken care of the CDR updates, and it would
be nicer to put the 2 SQL statements together).
I don't think you can use this in a high volume environment too, because of
the possible load increase on your database when running the runthis.sh.

But still I wanted to share this with you, I hope it may be helpful to you
people having problem with BRIStuff/CDR.

Jeroen


- Original Message - 
From: Jeroen Zwarts [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 09, 2006 11:05 AM
Subject: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x


 I have a problem with CDR recording in Asterisk 1.2.x. This is the
 situation:

 An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a
single
 HFC-S ISDN BRI card. I log the call records to both the Master.csv and
 MySQL.

 The problem is that when an incoming call from the ISDN line is logged to
 the CDR, the src and the clid field show up as something like 'h?'
 (random weird ASCII characters). This is in the MySQL table as well as the
 Master.csv, so my guess is that it is not a MySQL problem. Furthermore, I
 don't think it is a zaptel/bristuff problem, because my AGI scripts get
the
 incoming number without problems all the time.
 The internal SIP calls are logged without a problem all the time. It's
only
 ISDN calls from the outside world that are corrupt.


 When I stop Asterisk with stop now and restart it, the src and clid
 fields are OK for a while, but after a few calls, or as some time passes
by
 (I don't know what triggers it), it goes back to the 'random ASCII
 weirdness'.

 I also tested this with Asterisk 1.2.4 (BRIstuffed-0.3.0-PRE-1h with
florz)
 and I have the same problem. Again, when I start Asterisk, everything is
OK
 for a while, and then suddenly, the src and clid fields are like 'ÀÜ'

 Anybody has a clue as where to start looking for a solution for this
 problem? I can't seem to find a single post, list e-mail or bug related to
 this problem.

 Thanks,

 Jeroen Zwarts

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Re: [Asterisk-Users] Dumb question (hang up detection/Zapata.conf)

[artifex maximus]

 My asterisk system seems to have problems detecting hangups.  I am
 getting a LOT of voicemails with dialtone or silence.

 I am using an external gateway (wellgate 3701a) and don't have zaptel
 at all.
I think your 3701a don't understand hangup tone (as our 3802 did and
keep line busy after disconnect).

You need training the device. We had recorded the busy tone and
analyze with an audio editor (tone frequency and duration) and entered
the configuration via telnet interface (I think web interface is good
as well for this). And as a second chance we entered the values from
the suggested pdf in indications.conf:
http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf

bye,
Zsolt
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Re: [Asterisk-Users] Asterisk RealTime Question, Please help

[Benchev]

I was able to install Asterisk and Asterisk-addons and use them
successfully. But I have a problem now, I have many contexts and it
looks like Asterisk is unable to find the context given directly in
Mysql DB unless I specify it in Extensions.conf to switch it to
RealTime. If I add a new context in Mysql then I have to add it in
Extensions.conf and reload extensions whenever I need a new context.
Please tell me if there is a way to avoid all this and make Asterisk
take contexts directly from Mysql without mentioning that context in
Extensions.conf. If this is possible then I can make my Asterisk
RealTime actually and modify contexts directly in Mysql.
 
  Idea from the wiki:
  ; If regcontext is specified, Asterisk will dynamically create and
  destroy a ; NoOp priority 1 extension for a given peer who registers or
  unregisters with ; us.  The actual extension is the 'regexten' parameter
  of the registering ; peer or its name if 'regexten' is not provided. 
  More than one regexten may ; be supplied if they are separated by ''. 
  Patterns may be used in regexten. ;
  ;regcontext=sipregistrations
  That means that you should creat a mother context in extensions.conf:
  [sipregistrations]
 
  But first I would try to add a field regcontext along with
  regexten(which already there) in sip_users table since for the trick to
  work ... read http://www.voip-info.org/wiki-Asterisk+sip+regcontext
 
 OK, that will enable the auto generation of a context but as the new
 context won't have a switch statement it doesn't help with this
 problem... I may try writing a default switch if no matching context
 found type patch.
Well, it wont generate a context, it would rather register the extension of 
the new user under [sipregistrations]

And, maybe now is the time to warn that regexten was created to facilitate
a sip-user extensions' propagation within an * network; there is a 
discussion Clustering going on the list, see for details.

As for the switch, since context is optional:
 (switch = Realtime/@realtime_ext) and if left off, RealTime will use the 
current context, in this case sipregistrations.
Means:
[sipregistrations]
switch = Realtime/@realtime_ext ;realtime_ext or whatever the table name is

Ok i'am guessing sans voir here since I don't understand why so many
contexts are needed?
Hope it helps,
Benchev

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RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?

[Peter Braidwood]

My systems work perfectly with 8.2, hit dial from the missed calls menu
and the call is placed exactly as expected.

Peter

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Stenton
Sent: 14 March 2006 09:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?

Maybe I have something strange in my dial plan but I have no problem
just 
hitting dial from missed calls under 8.2.

Chris

- Original Message - 
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, March 13, 2006 8:44 PM
Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?


 We rolled back to 7.4 cause of that too.  7.5 has a strange bug where
if 
 the server loses connection, the phone's just don't try
re-registering.

 Aaron

 Tim Connolly wrote:
 Just curious, why not 7.5 ? -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
 Jafferali
 Sent: Monday, March 13, 2006 2:28 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?

 I'm using P0S3-08-2-00.. I noticed the callerID started showing
 up
 with the number, then @proxy-addr... So the callerID on the phone 
 looks like: [EMAIL PROTECTED] which of course is logged in the 
 missed calls exactly like that, and completely foobars the dialing 
 string if you try to dial a missed call by simply hitting the dial 
 button. Can anyone else verify this problem?

 Yeah, that bothered me so I rolled back to SIP 7.4.

 Nabeel

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Re: [Asterisk-Users] saydigits

[Rich Adamson]

Jerry Geis wrote:

I was searching on voip-info.org for saydigits.
I see no indication it is not valid in 1.2.4 asterisk.
however, when trying to use it I get and error no application saydigits.

what is the correct way to echo back digits in asterisk 1.2.4?

I tried say digits 123 and saydigits 123 both gave no application  
error


Try something like this as an experiment:
; Read back caller's number
exten = 3912,1,Wait(1)
exten = 3912,2,SayDigits(${CALLERID(num)})
exten = 3912,3,Hangup


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Re: [Asterisk-Users] Can One FXO Support Multiple Phone Lines?

[Rich Adamson]

Looks like the original question posed to the OP had to do with physical 
wiring, as in red/green equals line #1 and yellow-black equals line #2.



James Harper wrote:

Definitely one line per FXO port, but the wording of the original poster
was two numbers, not two lines, and while it may not be universally
true, distinctive ring should allow two (or more) phone numbers to be
present on an FXO port, and asterisk should be able to tell which one is
calling.

If the original poster did mean lines and not numbers, maybe there was
some confusion about the difference between PSTN and ISDN.

James


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Tuesday, 14 March 2006 12:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can One FXO Support Multiple Phone

Lines?

In a word...No.

One line per FXO port.

Next question?

Michael

--Original Message Text---
From: Andrew Berman
Date: Mon, 13 Mar 2006 19:32:29 -0500

I am currently having our new office wired up with 8 PSTN lines. The

guy

asked me if he could wire it up such that one line had two phone

numbers.

I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if all

I

needed were 4 FXO ports. Is it possible to set up Asterisk with 2

numbers

per FXO?

Thanks for any help,

Andrew


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Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5

[Matt]

Right saw that.   But I'm trying to get away from using CVS-HEAD :)  
Is the jitterbuffer patch PURELY 1.2.5 with the patch in place?


On 3/14/06, Olle E Johansson [EMAIL PROTECTED] wrote:

 13 mar 2006 kl. 21.59 skrev Matt:

  Hi,
  I really want to start using 1.2.5, but I also really need to have the
  jitter buffer.  Can anyone offer a suggestion of how to go?   I've
  looked at the SIP_JB patch and there seems to be no indication of a
  patch for the 1.2.5 release.

 Look again. There is a new branch called jitterbuffer-1.2 that follows
 svn HEAD in the 1.2 branch. This is documented in the bug tracker
 report for the jitterbuffer :-)

 Please test! Thanks!

 /O


 ---
 * Olle E. Johansson - [EMAIL PROTECTED]
 * Asterisk Training http://edvina.net/training/



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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

[Omar A. Sabek]

Chris, you may have a different and simpler setup. Internal calls work
fine here, since the proxy server on the CallerID is the same proxy
server used for all internal users. I was referring to calls that
originate outside of the enterprise. I should have been more clear.

Omar



On 3/14/06, Chris Stenton [EMAIL PROTECTED] wrote:
 Yes it does display caller id as callingnumber@ip of calling party but
 that does not interfere with me hitting dial from missed calls. Seems the
 Cisco phone sends the  sip INVITE as  callingnumber@ip of calling party
 rather than callingnumber@ip address of defaultproxyserver but asterisk
 ignores the info after the @?

 Chris

 - Original Message -
 From: Omar A. Sabek [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, March 13, 2006 11:45 PM
 Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time


  The 8.2 firmware displays Caller ID as callingnumber@proxyaddr...
  this becomes problematic for users that want to dial from their
  'Missed Calls' log.
 
  Omar
 
  On 3/13/06, Nathan Bowyer [EMAIL PROTECTED] wrote:
  On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote:
   I have had no issues with 8.2 so far!
  
   Chris
  
 
  Except the Caller ID issue reported in another thread?
 
   
This issue has been fixed in SIP firmware 7.5
   
Omar A. Sabek
   
Yes, and I read that SIP 7.5 firmware have some other issues. They
recommend using 7.4 firmware. I'm not sure how good in new 8.2
firmware.
   
   
Tomislav
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Re: [Asterisk-Users] RE: Delay in ringing

[Wireless]

I found that the Fax detect delay in the extentions.conf was causing my
system to have a delay

[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2)
; if fax detection is disabled, then jump to from-pstn-nofax - else continue
exten = s,2,Answer
exten = s,3,Wait(0)
exten = s,4,SetVar(intype=${INCOMING})
exten = s,5,Cut(intype=intype,-,1)
exten = s,6,GotoIf($[${intype} = EXT]?7:9) ; If INCOMING starts with EXT,
then assume its an extension
exten = s,7,Wait(0)

I changed s,7 to 0.  Not sure what will happen if a fax arrives... but then
I don't care about faxes

Harvey
- Original Message - 
From: Ash Thakrar [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, March 13, 2006 4:12 PM
Subject: [Asterisk-Users] RE: Delay in ringing



 Hi All,

 I am running [EMAIL PROTECTED] with Digium TDM400 card with FXO modules
plugged
 to PSTN lines.

 I am currently experiencing a delay in ringing by around 12 seconds.

 Is there something I need to adjust in the dial plan for this?

 Regards
 Ash

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[Asterisk-Users] Latest Dell SC430 Compatibility With Wildcard

[Leo Ann Boon]

Anyone knows if the SC430, based on the Intel E7230 chipset, is 
compatible with the Digium cards? I've tried the compatibility page on 
digium's website. It seems like they've pulled the old compatibility 
list, now the links on the page only point back to the product pages. 
Over here, Dell is selling (for a short period of time), SC430 with 
Pentium D 820 Dual Core Processor 2.8GHz, 256MB RAM, 80GB SATA for about 
US$240.





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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Tue, 14 Mar 2006, Omar A. Sabek wrote:


Chris, you may have a different and simpler setup. Internal calls work
fine here, since the proxy server on the CallerID is the same proxy
server used for all internal users. I was referring to calls that
originate outside of the enterprise. I should have been more clear.

Omar



Do you have canreinvite=no in the phone definition in sip.conf?

I am running our Cisco 7960s that way and under v8.2 the CallerID always 
shows the IP of the local asterisk server.  This way hitting the Dial 
softkey works perfectly wherever the call originated.


- -- 
Ron Wellsted

[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2.2 (GNU/Linux)

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3EzcyMqDG1Fgly4IxRFpML480TFN+cxqZAflFB92cwECO980y/geGN3XZA6izHKK
4PC+90iWCjhXFUR7aJo+wJ2jkCA/BozAQiGDA2wtkctRy0OQEdaAsxiRt5gY/Sm7
9xSz82KNXp0HM/InBK1abwd4n0UQ9Wm+v+3wrdD3XL0elp0FFQaaesSZS2PDMWCT
JSdDPfDoWN7t+VeDEeA+qugTYvt3HBJF8pDOzogg8Tnw1hhFXIYeATe8p2XNypkN
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Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?

[Aaron Daniel]

We only had the problem when the call was redirected from one server to 
another.  So if a phone was called from another phone on the server, the 
called worked perfectly, but if it was redirected from another server, 
we got the proxy added to the end.  Doesn't help when you're trying to 
make the existence of multiple servers transparent.


Aaron

Chris Stenton wrote:
Maybe I have something strange in my dial plan but I have no problem 
just hitting dial from missed calls under 8.2.


Chris

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

Sent: Monday, March 13, 2006 8:44 PM
Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?


We rolled back to 7.4 cause of that too.  7.5 has a strange bug where 
if the server loses connection, the phone's just don't try 
re-registering.


Aaron

Tim Connolly wrote:

Just curious, why not 7.5 ? -Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Monday, March 13, 2006 2:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?


I'm using P0S3-08-2-00.. I noticed the callerID started showing

up
with the number, then @proxy-addr... So the callerID on the phone 
looks like: [EMAIL PROTECTED] which of course is logged in the 
missed calls exactly like that, and completely foobars the dialing 
string if you try to dial a missed call by simply hitting the dial 
button. Can anyone else verify this problem?


Yeah, that bothered me so I rolled back to SIP 7.4.

Nabeel

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Re: [Asterisk-Users] Single E1 with HW Echo Can?

[Steve Davies]

Sangoma are about to release a 2-port card I believe, but I have not
heard of a 1-port unit. You would need to buy an external device,
which will probably raise to cost so close to the 2-port solution that
you may as well use that instead.

Regards,
Steve

On 3/9/06, Avi Miller [EMAIL PROTECTED] wrote:
 Hey guys,

 Is anyone aware of a single PRI (E1) card that has onboard hardware echo
 cancellation? I can only seem to find 4-port cards (Digium and Sangoma)
 and I really don't need the other three ports.

 Thanks,
 Avi

 --
 National Manager - Special Projects

  Sydney / Melbourne / Canberra / Hobart / London /
2/340 Gore Street  T: +61 (0) 3 9486 0411
Fitzroy, VIC   F: +61 (0) 3 9486 0611
3065   W: http://www.squiz.net/

 . Open Source  - Own it  -  Squiz.net ./
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Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5

[Olle E Johansson]

14 mar 2006 kl. 13.35 skrev Matt:


Right saw that.   But I'm trying to get away from using CVS-HEAD :)

We all are. Every developer have switched from CVS to Subversion :-)

This is not the development branch, but the release branch code,
which we use to create the 1.2.x releases.

The jitterbuffer itself is *not* release branch code, it's very much
development. Please test it.

The jitterbuffer branch is based on svn trunk (the same as the old  
CVS HEAD)
The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD  
(meaning latest 1.2 version code).


/O
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[Asterisk-Users] 10minutes to restart [EMAIL PROTECTED] 2.7

[Marco Mouta]

Hi all,

I've bought a TE110P, and received it today. So i decided to install
[EMAIL PROTECTED] 2.7 with this card.

In the past i had experiencies with X100P (clone card) and it never
take me so long to reboot the machine

Machine:
P4- 2,8Ghz 1GRAM
TE110P

What could be wrong?

Best regards,
Marco Mouta
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Re: [Asterisk-Users] Latest Dell SC430 Compatibility With Wildcard

[Time Bandit]

 Anyone knows if the SC430, based on the Intel E7230 chipset, is
 compatible with the Digium cards? I've tried the compatibility page on
 digium's website. It seems like they've pulled the old compatibility
 list, now the links on the page only point back to the product pages.
 Over here, Dell is selling (for a short period of time), SC430 with
 Pentium D 820 Dual Core Processor 2.8GHz, 256MB RAM, 80GB SATA for about
 US$240.
I have one * in production on this model. My spec are 1 gig RAM, 2 x
HD 160 gig (software-RAID) and a Digium TDM 2400P (8 FXO) with
hardware echo-canceller.

I would not say that it is the best system I have * running on, but it
does the job for that customer (4 person in a small office).

So, Yes, it is compatible.

hth
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Re: [Asterisk-Users] slinear bandwidth

[Andrew Kohlsmith]

On Monday 13 March 2006 23:16, Anton Krall wrote:
 Might be good for faxing though

Doubtful.  Faxes are designed to work within g711 limits.  I personally have 
been faxing through Asterisk (Canon and Xerox fax machines, the most 
notorious for being fickle) for well over a year now. It generally works.

My call path is this:

Fax - Adit600 - TE405 - Asterisk - 1-hop SDSL - Asterisk - TE405 - Telco

-A.
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[Asterisk-Users] Line connections

[Jeff Hoppe]

 Is there a way to connect a phone line to
another line that is in the offhook state?



 The Dial() application evidently needs to call
the other line (onhook state or a busy signal given), I would like the other
line to be already offhook and the phone line then gets connected.



Thanks

Jeff






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[Asterisk-Users] Problem with poud key (#)

[Matteo Piazza]

I not understand why my asterisk send the tone of pound key (#) only 
when i click twice time.

I deactivate the transfer function.
Matteo

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[Asterisk-Users] CDR question

[Marc Patino Gómez]

Hi,

i'm trying without success to change the dst (destination) entry of the 
cdr. I'm using the following:



exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)

I want to record into the cdr only the called number, but in the cdr 
appears the prefix 2006234500254.


I read this doc: 
http://digium-cvs.netmonks.ca/viewcvs.cgi/asterisk/doc/README.variables?rev=1.44


Can anybody help me?

Thanks in advance


--


Marc Patino Gómez
Dpto. Sistemas

Claranet España. Servicios Internet
C/General Almirante 2-28, Torres Cerdá
08014 Barcelona
Tel: +34 93 445 26 50
Fax: +34 93 445 19 20
www.claranet.es

Claranet Group: United Kingdom - Spain - France - Germany - Portugal - 
Netherlands - USA



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Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5

[Matt]

 The jitterbuffer branch is based on svn trunk (the same as the old
 CVS HEAD)
 The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD
 (meaning latest 1.2 version code).

Ok... so if I pull the 'jitterbuffer' branch I would get 'svn trunk code'.

But if I pull 'jitterbuffer-1.2' I get the same code as I would have
if I downloaded 1.2.5 and then applied a jitterbuffer patch (which I
know, does not exist for 1.2.5).

Is that correct?
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[Asterisk-Users] [OT?]SCCP image for cisco 7905g

[Simone Ricci]

Hi,
I recently purchased a brand new 7905g with his SIP firmware (licensed).
Now, I want to play a little with chan-sccp, but I'm unable to find the
appropriate firmware for my phone. I know that I must get it directly
from cisco, but before purchasing it will be very good for me to try a
bit; does someone can provide me (or even tell me where can I find) the
SCCP image (version doesn't matters as long it works with asterisk) for
the phone?

Thanks in advance,
Simone.
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Re: [Asterisk-Users] Development news :: T38 passthrough support

[James Sizemore]

The sipura 2100 does work good with a AS5300


Zoa wrote:


Does anybody know what devices really support t.38 ? I've seen a few 
claiming they do on the box, but most do not seem to support it at all.


Zoa.

Kristian Kielhofner wrote:


Olle E Johansson wrote:


Friends in the Asterisk.org community,

There is a lot of cool stuff going on in Asterisk development, 
things  that will change Asterisk and
make it work better in your organisation, make it easier to sell in  
your area or give you more
consulting oppurtunities - in short, functionality that will make a  
lot of sense for you users.


However, developers can't really get anywhere without a dialog with  
the users. You know
what you need, you know what is missing and how you would like to  
make Asterisk a better

choice.

I am planning to send out a description of new features now and  
then,  to inform you about
what is going on, but also to get some feedback. The bug tracker is  
not only a tool for developers,

but also for testers and users to react to changes and contribute.

*** ITU T.38 -- Fax over VoIP



Olle,

Let's say that I wanted to setup a complete environment to test 
this. I presume that I would need the following:


Fax machine
T.38 compliant ATA (Sipura claims this)
Asterisk server
T.38 compliant something - does this need to be a Cisco 5300 (or 
similar)?  Can it be just another plain ATA and fax machine?


Please suggest some possible hardware!

Thanks!



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Re: [Asterisk-Users] Real Time Asterisk

[Fernando Lujan]

Sean Cook wrote:

Yes you do need unixODBC before you compile asterisk.  Once you have
installed unixODBC , asterisk will compile and offer you the following
modules:

cdr_odbc.so  
res_config_odbc.so  
res_odbc.so


res_odbc.conf and cdr_odbc.conf are the related config files...


Now, I have asterisk working with realtime support.

But I'm having the following error:

*CLI realtime load extensions context internal
  Column Name  Column Value
   
   id  1
  context  internal
exten  611
 priority  1
  app  Playback()
  appdata  hello-world

When I try to call the 611 number.

*CLI Mar 14 11:51:16 NOTICE[8231]: pbx.c:1729 pbx_extension_helper: 
Cannot find extension context 'internal'


Someone can help me with this?

Thanks in advance.

Fernando Lujan

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[Asterisk-Users] Sample SER + Asterisk conf?

[Bart J. Smit]

I'm still having problems with ser and asterisk on the same public
server.

Could anybody send me a tarball of their ser.cfg and sip.conf off-list,
so I can do a sanity check against my files?

Much appreciated.

Bart...
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[Asterisk-Users] Realtime SIP

[Douglas Garstang]

Is anyone using realtime sip for friends (ie phones) with multiple Asterisk 
boxes all pointing to the one central MySQL database? Does it work? Are phones 
that are registered to the database from Asterisk box able to reach phones 
registered to the database from another Asterisk box?

Doug.

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Re: [Asterisk-Users] CDR question

[Benchev]

 i'm trying without success to change the dst (destination) entry of the
 cdr. I'm using the following:
 exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
 exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
 I want to record into the cdr only the called number, but in the cdr
 appears the prefix 2006234500254.
Would you try:
exten=_2006234500254.,2,Set(destination = ${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/[EMAIL PROTECTED],60,tr)

Benchev
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RE: [Asterisk-Users] RE: Predictive Dialer

[Adam Vocks]

Matt,

Without getting into a phone war...

What phones or headsets or softphones do you use with your installation?

Thanks

Adam

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Friday, March 10, 2006 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RE: Predictive Dialer

Hello,

I have used GnuDialer in a test environment and it does work. There
isn't much documentation out there on it but it is in production at
several sites. You should go to the GnuDialer website and post on
their forums for more information.
http://www.gnudialer.org/

The other GPL predictive dialer for Asterisk is VICIDIAL(which I am
the primary developer of) It is in production at over 100 companies
around the world and installs on top of almost any existing Asterisk
installation. Our company uses it for over 200 seats across 4
locations. The largest installation I know about is over 300 seats at
a financial services company. There are also many installations in
South and Central America and VICIDIAL is available fully translated
in Spanish.
http://astguiclient.sourceforge.net/vicidial.html

MATT---

On 3/10/06, Vladimir Montealegre [EMAIL PROTECTED] wrote:
 wath is the link of the vcidialer?

 Vladimir Montealegre Estailes
 Bogota-Colombia

 Este Mensaje Esta Hecho 100% con Electrones Reciclados
 - Original Message -
 From: Saul Diaz [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Friday, March 10, 2006 11:29 AM
 Subject: Re: [Asterisk-Users] RE: Predictive Dialer


  Adam Vocks wrote:
 
  OK, so apparently no one is using GnuDialer, is anyone out there using
  any other predictive dialers on asterisk?
 
 
  Thank you,
 
 
  Adam Vocks
 
 
  
 
  *From:* Adam Vocks
  *Sent:* Thursday, March 09, 2006 12:41 PM
  *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
  *Subject:* Predictive Dialer
 
 
  Hello all,
 
 
  I have a client interested in GnuDialer.  My question is:  Is there
  anyone on this list who has been using GnuDialer and I was wondering if
  you would be willing to share your experiences with it.
 
 
  Thank You
 
 
  Adam
 
 
 
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  I am using VCIDialer for testing purposes.. and work fine... 70 concurrent
  calls, a little heavy to install
 
  regards
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 Tutopia es Internet para todos.
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RE: [Asterisk-Users] Latest Dell SC430 Compatibility With Wildcard

[Kerry Garrison]

If you go into the BIOS and disable all unneeded devices (serial, parallel,
USB, floppy, etc) then you shouldn't have any problem. I have one in a 15
user setup that is working fine.
-Kerry
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Leo Ann Boon
 Sent: Tuesday, March 14, 2006 5:00 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Latest Dell SC430 Compatibility 
 With Wildcard
 
 Anyone knows if the SC430, based on the Intel E7230 chipset, 
 is compatible with the Digium cards? I've tried the 
 compatibility page on digium's website. It seems like they've 
 pulled the old compatibility list, now the links on the page 
 only point back to the product pages. 
 Over here, Dell is selling (for a short period of time), 
 SC430 with Pentium D 820 Dual Core Processor 2.8GHz, 256MB 
 RAM, 80GB SATA for about US$240.
 
 
 
 
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Re: [Asterisk-Users] Realtime SIP

[Aaron Daniel]

We were using this setup for a while (well, it was using odbc, but same 
concept).  What we did was configured the phones to register with all 
the servers basically, so each phone was reachable by each server, and 
if a phone didn't register with a server for some reason, we have 
mechanisms in place to send the call to another server to check there. 
Worked like a charm :)


Aaron

Douglas Garstang wrote:

Is anyone using realtime sip for friends (ie phones) with multiple Asterisk 
boxes all pointing to the one central MySQL database? Does it work? Are phones 
that are registered to the database from Asterisk box able to reach phones 
registered to the database from another Asterisk box?

Doug.

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Re: [Asterisk-Users] misdn

[asterisk]

Sorry for the trivial question I did
The answer is:

Only install linux kernel-default in Yast Software Management

Andrea




   
 [EMAIL PROTECTED] 
 .it   
 Sent by:   To 
 asterisk-users-bo asterisk-users@lists.digium.com 
 [EMAIL PROTECTED]  cc 
 m.com 
   Subject 
   [Asterisk-Users] misdn  
 13/03/2006 16.35  
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




Hi all,
I just  arrived in Italy from Cebit, qhere I spoke with digium and Beronet
people.
They told me to try to use the mISDN stack  to drive beronet and the new
upcoming digium ISDN Cards.

SO I searched, find
http://www.beronet.com/download/card_installation_guide.pdf, and I
immediately got the error:

asterisk01:~ # cd /usr/src/install-misdn/
asterisk01:/usr/src/install-misdn # make install
CONFIG_SMP=y


!!
Disable the SMP Setting in your Kernel Config.



make: *** [test_preempt] Error 1

So I discovered that mISDN does not support SMP and preempitive
multitasking.

but how can I disable this on my Suse Linux 10.0 box  ?
I found somemody saying make oldconfig, but I tryed it and rebooting the
pc I had no changes.

So what am I doing wrong ?

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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RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working

[Wai Wu]

Title: [Asterisk-Users] Clustering "NEW THREAD", Almost Working



Now, Iknow what 
you guys been talking about. It is likeDSN forsip phones, not really 
clustering. I original thought that you guys want to setup some thing that can 
fail over to a different sip server if the server running the IVR 
dies.


From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas 
GarstangSent: Tuesday, March 14, 2006 12:11 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: RE: 
[Asterisk-Users] Clustering "NEW THREAD", Almost 
Working

Holy crap. You got SIP realtime working? I've tried it twice before and it 
failed the same way twice. Do you have multiple Asterisk boxes accessing the 
same sip info (ie phones) in the same table on the same database? Digium has 
said numerous times this known not to work, although I cant' work out why as 
it's just reading from a common table.

  -Original Message- From: JR Richardson 
  [mailto:[EMAIL PROTECTED] Sent: Mon 3/13/2006 7:11 PM 
  To: asterisk-users@lists.digium.com Cc: 
  Subject: [Asterisk-Users] Clustering "NEW THREAD", Almost 
  Working
  All,I made some progress, but it seems the further I 
  go with clustering theharder things get. Hmmm, I guess if it were 
  easy, it would bedocumented..Anyhow, I have 1 * server as the 
  DUNDi peering master with a ttl=1. Theonly function of this server 
  is to lookup where other sip peers areregistered and forward that info on 
  to the requesting * server.I have 4 * servers accepting registrations 
  from sip users (phones). All thesip phone info is stored in a MySQL 
  database and being accessed through therealtime engine, and it works 
  great. A phone registers to a server and theserver checks the 
  database and if an entry is present, the * servers allowsthe phone to 
  register and dumps the sip phone into sip show peers, worksgreat. I 
  can take the sip entry out of the database and the phone will notresister 
  in realtime. Works great.Now the dial plan setup. All the 
  extension info is also in the MySQLdatabase, I have a switch statement in 
  the [siptest] context pointing to thedatabase for extension logic. 
  This also works great. All servers arepointing to the same data 
  source with all sip extensions in the databasestarting withexten = 
  1234,2,Answer and so onexten = 1235,2,Answer and so onnotice 
  the priority 2 starting point in the database, very important.This is 
  the good part, in sip.conf, I have regcontext=siptest in the 
  generalsection (because it doesn't work in the users section), so when a 
  sip phoneregisters on a server, * dynamically inputs an exten = 
  1234,1,Noop into thedialplan and immediately the phone is able to be 
  called. This is workingpretty damn well also.So at this 
  point I have several phones registered across 4 * servers, allpulling 
  their info from MySQL, the same data source. Now let's say phone1234 
  and 1235 are registered to server 1 and phone 1236 and 1237 areregistered 
  to server 2, 1234 can call 1235 and vise versa, 1236 can call1237 and vise 
  versa.Now from phone 1234 on server 1, I call 1236 on server 2 and 
  because 1236does not have a priority 1 entry on server 1, the call 
  progresses to a DUNDilookup statement in the diaplan logic and request 
  exten 1236 location fromthe DUNDi peering master server (these 
  registration servers all are peeredwith the dundi peering master server 
  with a ttl=2, so the request will getpast the peering master server and on 
  to the other registration servers).The request is answered from server 2 
  and 1234 can now complete a call to1236. This is great, all is well, life 
  is good, had a big Dallas barbequelunch to celebrate because all my sip 
  phones are dynamically registering toany one of 4 sip registration 
  servers, and the other three servers know whois registered where through 
  DUNDi lookups. And it only took me 2 weeks toget this 
  far.Now then, let's break it and see what happens, dial any sip phone 
  that isnot actively registered and you get an endless DUNDi lookup request 
  from allservers except the one you are dialing from. I only had one 
  other server onat this time and within seconds produced 590+ IAX trunks 
  initiated back intoa registration server before I could hang up the 
  line.As far as I can tell, if you make a call from server 1, exten 
  1234 to exten1236, but 1236 is not actively registered on any other 
  server, the otherserver will get the DUNDi lookup request and not know 
  where the phone is soit keeps looking up and calling itself to find an 
  extension that is notthere, or something, anyhow it's a bad 
  thing.Now intrinsically knowing that this protocol is smarter than me, 
  I'mguessing that I have incorrect dialplan logic that is allowing this 
  tohappen. I'm wondering how I can set up a dialplan flow that will 
  do this:From Server 1, pick up phone and dial a number 
  (phone)(exten),1. * checks to see if the 

Re: [Asterisk-Users] Realtime SIP

[Simon Woodhead]

Hi Doug,We use Realtime SIP via a central MySQL database (2 actually in Master  Master config) but registration is only available on the box to which the client has registered. Clients can register with any database and the table does get updated with some registration information (ip address, expiry time etc.) but they are not reachable by any of the other boxes sharing the config. I've been following the cluster thread with great interest for a workable solution to this.
All the best,SimonOn 3/14/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Is anyone using realtime sip for friends (ie phones) with multiple Asterisk boxes all pointing to the one central MySQL database? Does it work? Are phones that are registered to the database from Asterisk box able to reach phones registered to the database from another Asterisk box?
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[Asterisk-Users] Problem with uac_replace and corrupted From

[Barry Flanagan]

Hi,

Using openser 1.1.0-dev8 as a registrar/proxy in from of Asterisk. 
Recently I have been getting errors from Asterisk due to corrupted From: 
headers, which appear to be caused by uac_replace. Here is a section of 
the debug log:



Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: 
DBG:uac::restore_from_reply: removing From: 
sip:[EMAIL PROTECTED];tag=635c3ce6  
Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: 
DBG:uac::restore_from_reply: inserting From: 
sip:[EMAIL PROTECTED];tag=635c3ce6  
Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: DEBUG:tm:reply_received: 
org. status uas=200, uac[0]=200 local=0 is_invite=1)
Mar 14 15:12:00 www1 /usr/sbin/openser[7932]: DEBUG:uac:restore_from: 
getting 'vsf' Route param
Mar 14 15:12:00 www1 /usr/sbin/openser[7932]: DEBUG:uac:restore_from: 
Route param is 'aaafggqrsqadbekacyvdabuuaw1hz2luzs5pzq--' (len=48)
Mar 14 15:12:00 www1 /usr/sbin/openser[7932]: DEBUG:uac:restore_from: 
decoded uris are: new=[▒�S


 sN� old=[sip:[EMAIL PROTECTED]


As you can see, the auto restore_from of the uri is putting garbled 
characters in there. This has only recently started in the past week. I 
have made no changes to openser in a month. The only thing I did was 
upgrade Asterisk to 1.2.5


Anyone any ideas?
--

-Barry Flanagan



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[Asterisk-Users] digium.com redesign

[Michael George]

I may be way behind here, but I see that digium redesigned their site.
I cannot find the mailing list search screen.

I have found the mailman list page, but that doesn't have have a nice
search ability.

Do I need to just rely on google and other generic search engines or is
there a search on the digium site?

Thanks!
-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] Cannot load wcfxo -- Please help!

[Phil Freed]

At 11:32 AM 3/13/2006, John Daragon wrote:

Phil Freed wrote:
 I'm afraid that I am at a loss here.  I am new to Asterisk, and have
 successfully set up SIP.  But I cannot get my FXS card working, and I'm
 not sure what else I can try.

 # modprobe wcfxo

 /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: init_module: No such device

IIRC, wcfxo is the driver for the X100P card. The 4 port analog card's
driver used to be called wcfxs. but that led to the sort of confusion
you're experiencing, so it was renamed to wctdm.


Thank you for your response.  It's a big help to know where _not_ to 
look.  It turns out that the answer was right in front of me:


The system was purchased with the Digium card already installed; I assumed 
it was installed correctly.  And I had no documentation to go with it, so I 
had no idea that FXS was never going to work without a 12v source connected 
to the card.  (It makes sense, of course, but I never thought about it.)


So the answer was:  If all else fails, plug it in. :-)

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RE: [Asterisk-Users] digium.com redesign

[Ross C]

I prefer the google groups search

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael George
Sent: Tuesday, March 14, 2006 9:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] digium.com redesign

I may be way behind here, but I see that digium redesigned their site.
I cannot find the mailing list search screen.

I have found the mailman list page, but that doesn't have have a nice
search ability.

Do I need to just rely on google and other generic search engines or is
there a search on the digium site?

Thanks!
-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 20, Issue 91

[Noah Miller]

Hi Michael - 

 I may be way behind here, but I see that digium redesigned their site.
 I cannot find the mailing list search screen.

I don't believe there has ever been a search screen.


 Do I need to just rely on google and other generic search engines or is
 there a search on the digium site?

Do a google search for:

search terms site:lists.digium.com



- Noah

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Re: [Asterisk-Users] echo problem + choppy sound

[sdgesa gaeharth]

I have done this but I still get choppy sound and echo on some callsthanksGiovanni Miano [EMAIL PROTECTED] wrote:  Of course,Echo is 2 types: electric and ambiental.If u gain rx o tx more than you need, its return in recive and gen echoTry to decrase value, try to set 0 or .. in samecase -1 -2...  2006/3/13, sdgesa gaeharth [EMAIL PROTECTED]:  Can you explain why?Giovanni Miano [EMAIL PROTECTED] 
  wrote:rxgain=10.0  txgain=10.0  Maybe this is a problem2006/3/13, sdgesa gaeharth   [EMAIL PROTECTED]  :I  still hear a slight echo of my voice when I talk with somone out the  PSTN. The voice on the other end sounds very choppy and a little  distorted. When I talk to other people within our office, the sound is  perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: 
 [channels]  group = 1  language=encontext=incoming  signalling=fxs_ksswitchtype=national  usecallerid=yes  hidecallerid=no  callwaiting=yes  musiconhold=default  usecallingpres=yes  callwaitingcallerid=yes  threewaycalling=yestransfer=yes  canpark=yescancallforward=yes  callreturn=yes  echocancel=yes  echotraining=yes  echocancelwhenbridged=yes  rxgain=10.0  txgain=10.0  channel = 1-4  Yahoo! Mail  Bring photos to life!   New PhotoMail  makes sharing a breeze. ___--Bandwidth and Colocation provided by   Easynews.c   om --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano  ___--Bandwidth and Colocation provided by   Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users   
 Yahoo! Mail  Bring photos to life! New PhotoMail  makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.com   --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano  ___--Bandwidth and Colocation provided by Easynews.c
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RE: [Asterisk-Users] RFC Follow Me Find Me script

[Andrew Kirch]

Just poking this topic as it seems to have been ignored.  I still am not
clear as to how/where this script is broken.


If I read this correctly the syntax in column two is the current best
practice for AstDB.  It, unless I've missed something below is what I
have used in my script.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew D
Kirch
Sent: Friday, March 10, 2006 11:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RFC Follow Me Find Me script

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

(top posting to follow previous/keep thread sane)

* The applications DBGet and DBPut have been deprecated in favor of
  functions.  Here is a table of their replacements:

  DBGet(foo=family/key)Set(foo=${DB(family/key)})
  DBPut(family/key=${foo}) Set(DB(family/key)=${foo})

Johann wrote:
 That looks like the dialplan for Asterisk 1.0.x,  The AstDB and other
 commands have changed in Asterisk 1.2.x(and CVS HEAD).  Check the
 UPGRADE.txt in the source code directory of Asterisk to get the
details
 on all the changes...
 
 --johann
 
 Andrew D Kirch wrote:
 
 This is a follow/find me script that I can't quite get to work,
 asterisk wont save forward/${calleridnum} to AstDB... any comments or
 thoughts on how to make this work or change it to work differently
are
 appreciated.  The voice prompts to go with all playback/background
 extensions are commented appropriately.  I hope this code is of use
to
 some of you and any help with a perfected version is of course
 appreciated.
 [Forward]
exten = s,1,Playback(forward/extension-forwarding)
   ;Extension Forwarding
exten = s,2,GotoIf($[${CALLERIDNUM}300]?s,5)
   ;since 1xx is the pattern match for internal extensions anything
 less than 300 has to be internal so we already know that that is the
 extension they are wanting to forward
exten = s,3,Read(CALLERIDNUM,foward/please-ent-exten,3)
;if it's not have the user enter their 3 digit enternal extension
;please enter the extension you want to forward
exten = s,4,SayNumber(${CALLERIDNUM})
exten = s,5,Background(forward/extension-fwd-menu)
 ;to hear your current extension forward options press 1, to forward
 your phone press 2, to cancel your forwarding press 3


exten = 1,1,Set(FORWARD=${DB(forward/${CALLERIDNUM})})
exten = 1,2,NoOp(FORWARD is ${FORWARD})
exten = 1,3,GotoIf($[${FORWARD}0]?100,3)
exten = 1,4,Playback(forward/your-ext-not-forward)
;your extension is not currently forwarded
exten = 1,5,Goto(Forward,s,5)
;back to main menu
exten = 100,1,Playback(forward/your-ext-forward)
exten = 100,2,SayDigits(${FORWARD})
;your extension is currently forwarded to extension
exten = 100,3,Goto(Forward,s,5)
   ;back to main menu

exten = 2,1,Read(FORWARD,forward/please-ent-exten)
exten = 2,2,NoOp(FORWARD is ${FORWARD})
exten = 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD}
} )
exten = 2,4,NoOp(forward/${CALLERIDNUM} is ${
 DB(forward/${CALLERIDNUM} ) } )
exten = 2,5,Playback(forward/your-ext-forward-saved)
;your extension forward has been saved
exten = 2,6,Goto(Forward,s,5)

exten = 3,1,DBdel(forward/${CALLERIDNUM})
exten = 3,2,PlayBack(forward/exten-forward-cancel)
; your extension forward has been deleted.
exten = 3,3,Goto(Forward,s,1)

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[Asterisk-Users] Asterish Guru needed in Phoenix ASAP

[Kyle Hagan]

Need 2 full-time Asterisk Guru's needed in Phoenix area right away.
Knowledge in some of following also needed:
   Php
   Perl
   MySQL
   PostGres
   C++
   Visual Basic
   HTML
   Photo Shop
  
Email resume off list - Will interview this week.


Kyle

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[Asterisk-Users] Bad FXS Module?

[Jimmy]

I apparently have a dead FXS module.  Is there any kind of test I can 
run on it (on a live system) to determine if its good or bad?  The 
telephone has no dialtone, gets no calls. It has been working for 
several months, and just quit yesterday. Thanks!


Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
Freshmaker version: 73
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
ProSLIC on module 2 failed to powerup within 510 ms (0 mV only)

 -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P??
Unable to do INITIAL ProSLIC powerup on module 2
ProSLIC on module 2 failed to powerup within 510 ms (0 mV only)

 -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P??
Unable to do INITIAL ProSLIC powerup on module 2
Module 2: FAILED FXS (FCC)
Module 3: Installed -- AUTO FXS/DPO

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Re: [Asterisk-Users] CDR question

[Marc Patino Gómez]

I tried it, it didn't work, and I tried without success the following

exten=_2006234500254.,2,SetVar(destination = ${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/[EMAIL PROTECTED],60,tr)

Any help will be welcome

thanks

Benchev wrote:


i'm trying without success to change the dst (destination) entry of the
cdr. I'm using the following:
exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
I want to record into the cdr only the called number, but in the cdr
appears the prefix 2006234500254.
   


Would you try:
exten=_2006234500254.,2,Set(destination = ${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/[EMAIL PROTECTED],60,tr)

Benchev
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--


Marc Patino Gómez
Dpto. Sistemas

Claranet España. Servicios Internet
C/General Almirante 2-28, Torres Cerdá
08014 Barcelona
Tel: +34 93 445 26 50
Fax: +34 93 445 19 20
www.claranet.es

Claranet Group: United Kingdom - Spain - France - Germany - Portugal - 
Netherlands - USA



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Re: [Asterisk-Users] echo problem + choppy sound

[Rich Adamson]

Based only on what I see below (from previous posts), it sounds like you 
have two separate issues going on: 1) echo, and, 2) choppy sound. Those 
should be analyzed as two problems (not one).


You will find plenty of posts in the archives relative to both. In 
general terms, the choppy audio most often is caused by shared IRQ's 
when using a x100p or TDM400 card, and sometimes from a misconfigured 
ethernet nic on the asterisk machine. For the nic card, ensure you are 
running full duplex on the nic and whatever the nic is plugged in to.
Both need to be the same (half duplex will work in a low usage 
environment, but full duplex is preferred.)


For the IRQ issue (and we are all assuming you are using a TDM04b card 
since you really didn't say), do a 'cat /proc/interrupts' and make sure 
your TDM card is on its own IRQ. If it is shared with other devices, it 
is likely the cause for choppy audio. You'll see the TDM driver wctdm on 
that list. If it is shared, then move the TDM card from one pci slot to 
another to get it on its own IRQ.


The echo problem is going to be almost aways related to too high of 
gains in zapata.conf.  Your rxgain=10 and txgain=10 are way too high as 
others have already noted. Try reducing those to 0 and restart asterisk.
Then increase the values (if needed) by increments of 2 until you find a 
balance between low volume and echo.  I'd suggest doing that after 
resolving the IRQ/choppy audio issues.




I have done this but I still get choppy sound and echo on some calls

thanks

*/Giovanni Miano [EMAIL PROTECTED]/* wrote:

Of course,
Echo is 2 types: electric and ambiental.

If u gain rx o tx more than you need, its return in recive and gen echo

Try to decrase value, try to set 0 or .. in samecase -1 -2...

2006/3/13, sdgesa gaeharth [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:

Can you explain why?


*/Giovanni Miano [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]/* wrote:

rxgain=10.0
txgain=10.0

 

Maybe this is a problem


2006/3/13, sdgesa gaeharth  [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] :

I still hear a slight echo of my voice when I talk with
somone out the PSTN.  The voice on the other end sounds
very choppy and a little distorted. When I talk to other
people within our office, the sound is perfect.Can some
help?

We are all using:

Polycom 501 -- asterisk -- PSTN

zapata.conf:
[channels]
group = 1
language=en
context=incoming
signalling=fxs_ks
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
musiconhold=default
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=10.0
txgain=10.0
channel = 1-4



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Re: [Asterisk-Users] CDR question

[Benchev]

On Tuesday 14 March 2006 17:15, Benchev wrote:
  i'm trying without success to change the dst (destination) entry of the
  cdr. I'm using the following:
  exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
  exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
  I want to record into the cdr only the called number, but in the cdr
  appears the prefix 2006234500254.

 Would you try:
 exten=_2006234500254.,2,Set(destination = ${EXTEN:10})
 exten=_2006234500254.,3,Dial(OH323/[EMAIL PROTECTED],60,tr)
Yep fast end  stupid. Sorry, it wont work that way.

Benchev
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[Asterisk-Users] IAX choppy sound

[Stojan Sljivic - GDS]

Hi,

We have two Asterisk servers connected over IAX, with very limited bandwidth
256Kbs.
When we make calls between these two Asterisk servers the sound is very
choppy, no matter whether we use jitter buffer or not.

However, when we make calls using Skype, the sound is perfect.

Can anyone help us troubleshoot this IAX issue that we are experiencing?

Best regards,
Stojan Sljivic

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Re: [Asterisk-Users] Bad FXS Module?

[Rich Adamson]

Best bet... call digium support. The module should be under warranty 
(two years), and they will be able to tell you how to test if the notes

below aren't already enough.


Jimmy wrote:
I apparently have a dead FXS module.  Is there any kind of test I can 
run on it (on a live system) to determine if its good or bad?  The 
telephone has no dialtone, gets no calls. It has been working for 
several months, and just quit yesterday. Thanks!


Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
Freshmaker version: 73
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
ProSLIC on module 2 failed to powerup within 510 ms (0 mV only)

 -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P??
Unable to do INITIAL ProSLIC powerup on module 2
ProSLIC on module 2 failed to powerup within 510 ms (0 mV only)

 -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P??
Unable to do INITIAL ProSLIC powerup on module 2
Module 2: FAILED FXS (FCC)
Module 3: Installed -- AUTO FXS/DPO

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Re: [Asterisk-Users] CDR question

[Benchev]

  i'm trying without success to change the dst (destination) entry of the
  cdr. I'm using the following:
  exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
  exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
  I want to record into the cdr only the called number, but in the cdr
  appears the prefix 2006234500254.
But this should do it:
exten=_2006234500254.,2,Set(CDR(dst) = ${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)

Benchev
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Re: [Asterisk-Users] echo problem + choppy sound

[sdgesa gaeharth]

thanks for the info.it is not sharing an irq: 0: 59840409 59803082 IO-APIC-edge timer   8:  1  0 IO-APIC-edge rtc   9:  0 0  IO-APIC-level acpi  11:  0 0  IO-APIC-level ohci_hcd:usb1  14: 2141851 2143209 IO-APIC-edge ide0  177: 111558 111273 IO-APIC-level aic7xxx  185: 
 15 0  IO-APIC-level aic7xxx  193: 736328 748953 IO-APIC-level eth0  201: 239290099 239259220 IO-APIC-level wctdm  NMI: 0 0  LOC: 119645889 119645888  ERR: 0  MIS: 0  I checked the switch. The net connection is running at full duplex:FastEthernet0/15 is up, line protocol is up (connected)   Hardware is Fast Ethernet, address is 0013.80b7.e24f (bia 0013.80b7.e24f)   MTU 1500 bytes, BW 10 Kbit, DLY 100 usec,   reliability 255/255, txload 1/255, rxload 1/255   Encapsulatio
 n ARPA,
 loopback not set   Keepalive set (10 sec)   Full-duplex, 100Mb/s, media type is 100BaseTX   input flow-control is unsupported output flow-control is unsupported   ARP type: ARPA, ARP Timeout 04:00:00   Last input never, output 00:00:03, output hang never   Last clearing of "show interface" counters never   Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0   Queueing strategy: fifo   Output queue: 0/40 (size/max)   5 minute input rate 31000 bits/sec, 15 packets/sec   5 minute output rate 32000 bits/sec, 15 packets/sec   679924 packets input, 225898296 bytes, 0 no buffer   Received 3803 broadcasts (0 multicast)   0 runts, 0 giants, 0 throttles   0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored   0 watchdog, 
 5
 multicast, 0 pause input   0 input packets with dribble condition detected   689110 packets output, 145860377 bytes, 0 underruns   0 output errors, 0 collisions, 2 interface resets   0 babbles, 0 late collision, 0 deferred   0 lost carrier, 0 no carrier, 0 PAUSE output   0 output buffer failures, 0 output buffers swapped outRich Adamson [EMAIL PROTECTED] wrote:  Based only on what I see below (from previous posts), it sounds like you have two separate issues going on: 1) echo, and, 2) choppy sound. Those should be analyzed as two problems (not one).You will find plenty of posts in the archives relative to both. In general terms, the choppy audio m
 ost
 often is caused by shared IRQ's when using a x100p or TDM400 card, and sometimes from a misconfigured ethernet nic on the asterisk machine. For the nic card, ensure you are running full duplex on the nic "and" whatever the nic is plugged in to.Both need to be the same (half duplex will work in a low usage environment, but full duplex is preferred.)For the IRQ issue (and we are all assuming you are using a TDM04b card since you really didn't say), do a 'cat /proc/interrupts' and make sure your TDM card is on its own IRQ. If it is shared with other devices, it is likely the cause for choppy audio. You'll see the TDM driver wctdm on that list. If it is shared, then move the TDM card from one pci slot to another to get it on its own IRQ.The echo problem is going to be almost aways related to "too high" of gains in zapata.conf.  Your rxgain=10 and txgain=10 are way too high as others have already noted. Try reducing 
 those to
 0 and restart asterisk.Then increase the values (if needed) by increments of 2 until you find a balance between low volume and echo.  I'd suggest doing that "after" resolving the IRQ/choppy audio issues. I have done this but I still get choppy sound and echo on some calls  thanks  */Giovanni Miano /* wrote:  Of course, Echo is 2 types: electric and ambiental.  If u gain rx o tx more than you need, its return in recive and gen echo  Try to decrase value, try to set 0 or .. in samecase -1 -2...  2006/3/13, sdgesa gaeharth  :  Can you explain why?   */Giovanni Miano  /* wrote: 
 t;  
   rxgain=10.0 txgain=10.0     Maybe this is a problem   2006/3/13, sdgesa gaeharth  [EMAIL PROTECTED]  :  I still hear a slight echo of my voice when I talk with somone out the PSTN.  The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?  We are all using:  Polycom 501 -- asterisk -- PSTN  zapata.conf: [channels] group = 1 language=en context=incoming&
 gt; 
signalling=fxs_ks switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes 

[Asterisk-Users] Echo Cancellation

[Keith Schmidt]

I have 3 POTS lines that I want to use with Asterisk, I am looking at 
prices for FXO cards and the cards with echo cancellation are really 
pricey... is echo cancellation really worth it for a 3 or 4 line 
system?  Will I notice a difference without the echo cancellation?


Thanks
Keith Schmidt
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RE: [Asterisk-Users] RFC Follow Me Find Me script

[Jason Adams]

Andrew,

Don't know if this helps your or not, but it seems like you have one too
many {} in your set statement...
You have: Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
Try: Set(DB(forward/${CALLERIDNUM}=${FORWARD}))

 - Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kirch
Sent: Tuesday, March 14, 2006 11:21 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] RFC Follow Me Find Me script

Just poking this topic as it seems to have been ignored.  I still am not
clear as to how/where this script is broken.


If I read this correctly the syntax in column two is the current best
practice for AstDB.  It, unless I've missed something below is what I
have used in my script.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew D
Kirch
Sent: Friday, March 10, 2006 11:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RFC Follow Me Find Me script

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

(top posting to follow previous/keep thread sane)

* The applications DBGet and DBPut have been deprecated in favor of
  functions.  Here is a table of their replacements:

  DBGet(foo=family/key)Set(foo=${DB(family/key)})
  DBPut(family/key=${foo}) Set(DB(family/key)=${foo})

Johann wrote:
 That looks like the dialplan for Asterisk 1.0.x,  The AstDB and other 
 commands have changed in Asterisk 1.2.x(and CVS HEAD).  Check the 
 UPGRADE.txt in the source code directory of Asterisk to get the
details
 on all the changes...
 
 --johann
 
 Andrew D Kirch wrote:
 
 This is a follow/find me script that I can't quite get to work, 
 asterisk wont save forward/${calleridnum} to AstDB... any comments or

 thoughts on how to make this work or change it to work differently
are
 appreciated.  The voice prompts to go with all playback/background 
 extensions are commented appropriately.  I hope this code is of use
to
 some of you and any help with a perfected version is of course 
 appreciated.
 [Forward]
exten = s,1,Playback(forward/extension-forwarding)
   ;Extension Forwarding
exten = s,2,GotoIf($[${CALLERIDNUM}300]?s,5)
   ;since 1xx is the pattern match for internal extensions anything 
 less than 300 has to be internal so we already know that that is the 
 extension they are wanting to forward
exten = s,3,Read(CALLERIDNUM,foward/please-ent-exten,3)
;if it's not have the user enter their 3 digit enternal extension
;please enter the extension you want to forward
exten = s,4,SayNumber(${CALLERIDNUM})
exten = s,5,Background(forward/extension-fwd-menu)
 ;to hear your current extension forward options press 1, to forward 
 your phone press 2, to cancel your forwarding press 3


exten = 1,1,Set(FORWARD=${DB(forward/${CALLERIDNUM})})
exten = 1,2,NoOp(FORWARD is ${FORWARD})
exten = 1,3,GotoIf($[${FORWARD}0]?100,3)
exten = 1,4,Playback(forward/your-ext-not-forward)
;your extension is not currently forwarded
exten = 1,5,Goto(Forward,s,5)
;back to main menu
exten = 100,1,Playback(forward/your-ext-forward)
exten = 100,2,SayDigits(${FORWARD})
;your extension is currently forwarded to extension
exten = 100,3,Goto(Forward,s,5)
   ;back to main menu

exten = 2,1,Read(FORWARD,forward/please-ent-exten)
exten = 2,2,NoOp(FORWARD is ${FORWARD})
exten = 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD}
} )
exten = 2,4,NoOp(forward/${CALLERIDNUM} is ${ 
 DB(forward/${CALLERIDNUM} ) } )
exten = 2,5,Playback(forward/your-ext-forward-saved)
;your extension forward has been saved
exten = 2,6,Goto(Forward,s,5)

exten = 3,1,DBdel(forward/${CALLERIDNUM})
exten = 3,2,PlayBack(forward/exten-forward-cancel)
; your extension forward has been deleted.
exten = 3,3,Goto(Forward,s,1)

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[Asterisk-Users] Asterisk Users Group Meeting March 16, Irvine, Ca

[Kerry Garrison]

Irvine California, Heritage Park Library on the 
corner of Yale and Walnut. The Walnut is just south of the 5 fwy and Yale is 
between the Culver and Jeffery offramps. Meeting will run from 6 - 9pm. This 
week will feature a review of the SNOM 320, a demo of SIPX, some book giveaways 
courtesy of O'Rielly, and much more. 

For more information, contact me
Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com 


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[Asterisk-Users] Codec Issue

[Aisling]

Hi,
 
I have an issue which is kind of a catch 22 situation. I had outgoing calls
to my new PSTN provider working perfectly. Then I started focussing on
incoming calls. It seems that I can solve an error which gets my incoming
calls working but that in turns means my outgoing calls don't work. -
Strange
 
Anyhow I was getting an error: 
 
Process_sdp: No compatible codecs! 
And from the SIP debug I could see that the incoming SIP INVITE was getting
a sip response of 488 Unacceptable here from my asterisk server. 
 
After doing a bit of searching I determined that this might be the fault of
the codec's particularly the G729 codec. So in the peer block that I have
for my PSTN provider in my sip conf I specified allow=g729.
I called my PSTN geographic number again and was delighted when the incoming
calls worked. However when I next went to make an outgoing call (after
having added in the allow=g729 line), I got an infinite loop of warnings:
 
WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 256, while
native formats is 8 (read/write = 8/8)
WARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isn't a
multiple of 33 or 65 bytes long from RTP
 
After those warnings I thought there might be a problem with the gsm codec
so I commented the lines containing allow=gsm and still kept the line
allow=g729 because as I've said already incoming calls won't work
otherwise 9but outgoing will).
This however just gave another warning:
 
WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 4 while
native formats is 256 (read/write=64/64).
When I comment this line out again I am back to my original situation where
outgoing calls work and incoming don't.
 
I have included my sip.conf code and extensions.conf code below:
 
;sip.conf
 
[general]
bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
canreinvite=no;
autocreatepeer=yes
nat=yes
;dtmfmode=info
;dtmfmode=rfc2833
insecure=very
registerattempts=0
 
;context=default
 
register = [EMAIL PROTECTED]/1234
 
;To make outgoing calls specify this block
 
[providerIP]
type=peer
user=phone
host=providerIP
port=6060
fromdomain=providerIP
fromuser=username
secret=password
username=username
insecure=very
context=incomingpstn
authname=username
allow=gsm
allow=ulaw
allow=alaw
;allow=g729 ;NBNB This is where the issue is
 
[314]
type=friend
username=314
canreinvite=no
context=from-provider
insecure=very
host=dynamic
nat=yes
dtmfmode=rfc2833
mailbox=314
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
 
[2092]
type=friend
username=2092
canreinvite=no
context=from-provider
insecure=very
host=dynamic
nat=yes
dtmfmode=rfc2833
mailbox=2092
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
 
 
;extensions.conf
 
[general]
 
static=yes
writeprotect = yes
allow=alaw
 
;specify context for receiving incoming calls
 
[from-provider]
 
include = createmenu
include = createconf
include = joinconf
include = playvoicemail
;include = internalExt
;include = incomingpstn
include = default
 
[createmenu]
;Create an IVR Menu
exten = 20005,1,Wait(2)
exten = 20005,2,Record(/tmp/asterisk-recording:gsm)
exten = 20005,3,Wait(2)
exten = 20005,4,Playback(/tmp/asterisk-recording)
exten = 20005,5,wait(2)
exten = 20005,6,Hangup
 
[createconf]
;Create a conference call
exten = 20006,1,Wait(1)
exten = 20006,2,MeetMe(|MD)
exten = 20006,3,Hangup
 
[joinconf]
;Join a conference call
exten = 20007,1,Answer
exten = 20007,2,Wait(1)
exten = 20007,3,MeetMe(|P)
 
[playvoicemail]
;listen to voicemails
exten = 171,1,VoicemailMain(${CALLERIDNUM})
 
;Send PSTN calls to Provider
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup
 
[default]
 
;voicemail
exten = 314, 1,Dial(SIP/314,20)
exten = 314, 2,Voicemail(u314)
exten = 314, 102,Voicemail(b314)
exten = 314, 103,Hangup
 
exten = 2092, 1,Dial(SIP/2092,20)
exten = 2092, 2,Voicemail(u2092)
exten = 2092, 102,Voicemail(b2092)
exten = 2092, 103,Hangup
 
[incomingpstn]
 
;The below two lines dial a particular extension 
exten = 4590124,1,Wait(1)
exten = 4590124,n,Dial(SIP/[EMAIL PROTECTED],20,r)
 
 
 
 
 
 


---Legal  Disclaimer---

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contact the sender at the above quoted email address. Any unauthorised form of 
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Asterisk-Users 

[Asterisk-Users] Voip-Info

[Douglas Garstang]

Is it 
just me or is the voip-info web site down right now? Jeez that web site is 
flaky.

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Re: [Asterisk-Users] Codec Issue

[Rusty Dekema]

Hello. I am following the directions in your legal disclaimer because
I received a copy of your message, yet the message was not addressed
to me. My e-mail address is [EMAIL PROTECTED], but the message I
received was addressed to [EMAIL PROTECTED]

I would hate to see your confidential electronic mail transmissions
fall into the wrong hands, so I suggest that you take steps to ensure
that your confidential transmissions only reach their intended
recipients.

I hope that the reproduction of part of your message involved in
following the directions to notify you does not constitute
unauthorized reproduction, as I would hate to incur legal liability as
a result of your sloppy e-mail procedures.

Sincerely,
Russell Dekema
[EMAIL PROTECTED]


 ---Legal  Disclaimer---

 The above electronic mail transmission is confidential and intended only for 
 the person to whom it is addressed. Its contents may be protected by legal 
 and/or professional privilege. Should it be received by you in error please 
 contact the sender at the above quoted email address. Any unauthorised form 
 of reproduction of this message is strictly prohibited. The Institute does 
 not guarantee the security of any information electronically transmitted and 
 is not liable if the information contained in this communication is not a 
 proper and complete record of the message as transmitted by the sender nor 
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Re: [Asterisk-Users] Voip-Info

[Dave Cotton]

On Tue, 2006-03-14 at 10:21 -0700, Douglas Garstang wrote:
 Is it just me or is the voip-info web site down right now? Jeez that
 web site is flaky.

Is it just me or was this message in HTML? Jeez some people never learn.


Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] Attended Transfer - transfer timeout, how to change?

[Barry Flanagan]

Hi,

We are trying to use attended transfer with Asterisk 1.2.5, but when we 
do the transfer and dial the new number, it times out after 3 rings and 
then the callee is put back to the original agent.


Where can I adjust the timeout which applies to the number we are 
transferring to? I have changed the extension for this number to timeout 
at 60 seconds, but that seems to make no difference.


--

-Barry Flanagan
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[Asterisk-Users] OT - force Cisco phones to reboot

[Joao Pereira]

Hello to all
Does someone knows how to force the Cisco IP phones (7940 and 7960) to 
reboot weekly or monthly?
I think this would be useful because sometimes we change the 
configuration settings in the TFTP, but the phone just check the TFTP 
when he restarts...


Thanks
João
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RE: [Asterisk-Users] Voip-Info

[Douglas Garstang]

Oh, I'm sorry. I must have missed the previous message where you specifically 
informed me not to use HTML.

-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 14, 2006 10:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voip-Info


On Tue, 2006-03-14 at 10:21 -0700, Douglas Garstang wrote:
 Is it just me or is the voip-info web site down right now? Jeez that
 web site is flaky.

Is it just me or was this message in HTML? Jeez some people never learn.


Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Echo Cancellation

[Martin Joseph]

On Mar 14, 2006, at 9:04 AM, Keith Schmidt wrote:

I have 3 POTS lines that I want to use with Asterisk, I am looking at 
prices for FXO cards and the cards with echo cancellation are really 
pricey... is echo cancellation really worth it for a 3 or 4 line 
system?  Will I notice a difference without the echo cancellation?


This depends greatly on the quality of your PSTN line and the distance 
from the CO (central office).  In my case,  with a two wire loop over 
15000 feet,  I definitely had echo issues that made cheapo FXO 
unusable.


Although with a Digium card, you also have the option of using software 
based echo cancellation.  I have no experience with that.


Good echo cancellation is worth it in my opinion.
Marty

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RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working

[JR Richardson]

Yes, SIP realtime is working with multiple * servers all accessing the same 
MySQL database, add a sip phone in the database and the phone can register with 
any server without the need to configure any server, just add the phone in the 
database, petty cool. 

JR

--

Message: 21
Date: Tue, 14 Mar 2006 10:18:06 -0500
From: Wai Wu [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

Now, I know what you guys been talking about. It is like DSN for sip
phones, not really clustering. I original thought that you guys want to
setup some thing that can fail over to a different sip server if the
server running the IVR dies.
 
 
 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, March 14, 2006 12:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working


Holy crap. You got SIP realtime working? I've tried it twice before and
it failed the same way twice. Do you have multiple Asterisk boxes
accessing the same sip info (ie phones) in the same table on the same
database? Digium has said numerous times this known not to work,
although I cant' work out why as it's just reading from a common table.


JR Richardson
Engineering for the Masses

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Re: [Asterisk-Users] OT - force Cisco phones to reboot

[Steve Blair]

We send a notify message with the check sync event type. Not pretty but 
it works.


Joao Pereira wrote:


Hello to all
Does someone knows how to force the Cisco IP phones (7940 and 7960) to 
reboot weekly or monthly?
I think this would be useful because sometimes we change the 
configuration settings in the TFTP, but the phone just check the TFTP 
when he restarts...


Thanks
João
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--
 
ISC Network Engineering

The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  



voice: 215-573-8396 


  215-746-8001

fax: 215-898-9348


sip:[EMAIL PROTECTED]

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Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5

[Robert Webb]

On Tue, 14 Mar 2006 14:32:02 +0100
 Olle E Johansson [EMAIL PROTECTED] wrote:


14 mar 2006 kl. 13.35 skrev Matt:

Right saw that.   But I'm trying to get away from using 
CVS-HEAD :)
We all are. Every developer have switched from CVS to 
Subversion :-)


This is not the development branch, but the release 
branch code,

which we use to create the 1.2.x releases.

The jitterbuffer itself is *not* release branch code, 
it's very much

development. Please test it.

The jitterbuffer branch is based on svn trunk (the 
same as the old  CVS HEAD)
The jitterbuffer-1.2 branch is based on the 1.2 branch 
HEAD  (meaning latest 1.2 version code).


/O


Olle,

  Pardon this dumb question please, but where are these 
test located. I looked under http://svn.digium.com and do 
not see them. I am not fluent in where everything is 
located and would like to do some testing on some of the 
other items such as the sip jitterbuffer. It will only be 
minimal but I would like to help where I can.


Robert
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Re: [Asterisk-Users] OT - force Cisco phones to reboot

[Aaron Daniel]

It really depends on the number of phones you're wanting to reboot. 
Whenever we do a reconfiguration of our phones, I have a script that 
runs that night that pulls all the names from the db that are cisco 
phones, and does a sip notify cisco-check-cfg exten in asterisk, which 
notifies the phone to reboot in 20 seconds if nothing interesting 
happens (phone call comes in... browsing the interface... stuff like 
that).  In order for this to work, you have to put a file in the 
tftpboot folder called syncinfo.xml containing this:


SYNCINFO
IMAGE VERSION=* SYNC=0/
/SYNCINFO

in order for the phones to actually reboot though.

That's what we do anyway :)

Aaron

Joao Pereira wrote:

Hello to all
Does someone knows how to force the Cisco IP phones (7940 and 7960) to 
reboot weekly or monthly?
I think this would be useful because sometimes we change the 
configuration settings in the TFTP, but the phone just check the TFTP 
when he restarts...


Thanks
João
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Re: [Asterisk-Users] echo problem + choppy sound

[Rich Adamson]

Well... the next step (for me anyway) would be to use Ethereal on the 
asterisk nic interface to ensure the sip/rtp pkts are reasonable (eg, no 
dropouts). If those pkts flow consistently in both directions, then 
there must be something impacting the wctdm interface.


Do sip to sip calls sound reasonable?
Is there anything else running on your asterisk box?

sdgesa gaeharth wrote:

thanks for the info.

it is not sharing an irq:

  0:   59840409   59803082IO-APIC-edge  timer
  8:  1  0IO-APIC-edge  rtc
  9:  0  0   IO-APIC-level  acpi
 11:  0  0   IO-APIC-level  ohci_hcd:usb1
 14:21418512143209IO-APIC-edge  ide0
177: 111558 111273   IO-APIC-level  aic7xxx
185: 15  0   IO-APIC-level  aic7xxx
193: 736328 748953   IO-APIC-level  eth0
201:  239290099  239259220   IO-APIC-level  wctdm
NMI:  0  0
LOC:  119645889  119645888
ERR:  0
MIS:  0


I checked the switch.  The net connection is running at full duplex:

FastEthernet0/15 is up, line protocol is up (connected)
  Hardware is Fast Ethernet, address is 0013.80b7.e24f (bia 0013.80b7.e24f)
  MTU 1500 bytes, BW 10 Kbit, DLY 100 usec,
 reliability 255/255, txload 1/255, rxload 1/255
  Encapsulatio n ARPA, loopback not set
  Keepalive set (10 sec)
  Full-duplex, 100Mb/s, media type is 100BaseTX
  input flow-control is unsupported output flow-control is unsupported
  ARP type: ARPA, ARP Timeout 04:00:00
  Last input never, output 00:00:03, output hang never
  Last clearing of show interface counters never
  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
  Queueing strategy: fifo
  Output queue: 0/40 (size/max)
  5 minute input rate 31000 bits/sec, 15 packets/sec
  5 minute output rate 32000 bits/sec, 15 packets/sec
 679924 packets input, 225898296 bytes, 0 no buffer
 Received 3803 broadcasts (0 multicast)
 0 runts, 0 giants, 0 throttles
 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
 0 watchdog, 5 multicast, 0 pause input
 0 input packets with dribble condition detected
 689110 packets output, 145860377 bytes, 0 underruns
 0 output errors, 0 collisions, 2 interface resets
 0 babbles, 0 late collision, 0 deferred
 0 lost carrier, 0 no carrier, 0 PAUSE output
 0 output buffer failures, 0 output buffers swapped out

*/Rich Adamson [EMAIL PROTECTED]/* wrote:

Based only on what I see below (from previous posts), it sounds like
you
have two separate issues going on: 1) echo, and, 2) choppy sound. Those
should be analyzed as two problems (not one).

You will find plenty of posts in the archives relative to both. In
general terms, the choppy audio m ost often is caused by shared IRQ's
when using a x100p or TDM400 card, and sometimes from a misconfigured
ethernet nic on the asterisk machine. For the nic card, ensure you are
running full duplex on the nic and whatever the nic is plugged in to.
Both need to be the same (half duplex will work in a low usage
environment, but full duplex is preferred.)

For the IRQ issue (and we are all assuming you are using a TDM04b card
since you really didn't say), do a 'cat /proc/interrupts' and make sure
your TDM card is on its own IRQ. If it is shared with other devices, it
is likely the cause for choppy audio. You'll see the TDM driver
wctdm on
that list. If it is shared, then move the TDM card from one pci slot to
another to get it on its own IRQ.

The echo problem is going to be almost aways related to too high of
gains in zapata.conf. Your rxgain=10 and txgain=10 are way too high as
others have already noted. Try reducing those to 0 and restart asterisk.
Then increase the values (if needed) by increments of 2 until you
find a
balance between low volume and echo. I'd suggest doing that after
resolving the IRQ/choppy audio issues.


  I have done this but I still get choppy sound and echo on some calls
 
  thanks
 
  */Giovanni Miano /* wrote:
 
  Of course,
  Echo is 2 types: electric and ambiental.
 
  If u gain rx o tx more than you need, its return in recive and
gen echo
 
  Try to decrase value, try to set 0 or .. in samecase -1 -2...
 
  2006/3/13, sdgesa gaeharth
  :
 
  Can you explain why?
 
 
  */Giovanni Miano
  /* wrote:
 
g t; rxgain=10.0
  txgain=10.0
 
  
 
  Maybe this is a problem
 
 
  2006/3/13, sdgesa gaeharth  [EMAIL PROTECTED]
  :
 
  I still hear a slight echo of my voice when I talk with
  somone out the PSTN. The voice on the other end sounds
  very choppy and a little distorted. When I talk to other
  people within our office, the sound is perfect.Can some
  help?
 
  We are all using:
 
  Polycom 501 -- asterisk -- PSTN
 
  

Re: [Asterisk-Users] RE: Predictive Dialer

[Matt Florell]

Hello,

We use mostly Channelbanks and cheap analog phones with nice headsets.
Much cheaper in the long run and much easier/faster to replace the
phones. We also use Sipura ATA adapters with cheap analog phones and
nice headsets, and for our remote/external agents we use Firefly
third-party(free IAX softphone) with computer phone headsets.

The headsets we mostly use are Panasonic KX-TCA60 which you can
usually buy in bulk for under $15/each and are quite good quality. The
analog phones you can pickup all over the place at most discount
retail stores. As for Channelbanks we use mostly reconditioned Zhone
Zplexes that cost less than $300, they are small and quite reliable.

Hope that helps,

MATT---


On 3/14/06, Adam Vocks [EMAIL PROTECTED] wrote:
 Matt,

 Without getting into a phone war...

 What phones or headsets or softphones do you use with your installation?

 Thanks

 Adam

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell
 Sent: Friday, March 10, 2006 1:51 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] RE: Predictive Dialer

 Hello,

 I have used GnuDialer in a test environment and it does work. There
 isn't much documentation out there on it but it is in production at
 several sites. You should go to the GnuDialer website and post on
 their forums for more information.
 http://www.gnudialer.org/

 The other GPL predictive dialer for Asterisk is VICIDIAL(which I am
 the primary developer of) It is in production at over 100 companies
 around the world and installs on top of almost any existing Asterisk
 installation. Our company uses it for over 200 seats across 4
 locations. The largest installation I know about is over 300 seats at
 a financial services company. There are also many installations in
 South and Central America and VICIDIAL is available fully translated
 in Spanish.
 http://astguiclient.sourceforge.net/vicidial.html

 MATT---

 On 3/10/06, Vladimir Montealegre [EMAIL PROTECTED] wrote:
  wath is the link of the vcidialer?
 
  Vladimir Montealegre Estailes
  Bogota-Colombia
 
  Este Mensaje Esta Hecho 100% con Electrones Reciclados
  - Original Message -
  From: Saul Diaz [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Friday, March 10, 2006 11:29 AM
  Subject: Re: [Asterisk-Users] RE: Predictive Dialer
 
 
   Adam Vocks wrote:
  
   OK, so apparently no one is using GnuDialer, is anyone out there using
   any other predictive dialers on asterisk?
  
  
   Thank you,
  
  
   Adam Vocks
  
  
   
  
   *From:* Adam Vocks
   *Sent:* Thursday, March 09, 2006 12:41 PM
   *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
   *Subject:* Predictive Dialer
  
  
   Hello all,
  
  
   I have a client interested in GnuDialer.  My question is:  Is there
   anyone on this list who has been using GnuDialer and I was wondering if
   you would be willing to share your experiences with it.
  
  
   Thank You
  
  
   Adam
  
  
  
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   I am using VCIDialer for testing purposes.. and work fine... 70 concurrent
   calls, a little heavy to install
  
   regards
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RE: [Asterisk-Users] echo problem + choppy sound

[McQuiggan, Mark xt46480]

The best way to set gains would be to use ztmonitor (located in
/usr/src/zaptel).  Make a call and note your channel number.  Run
/usr/src/zaptel/ztmonitor channel number -v  from a telnet session.  check
to see if your levels are too high or too low and adjust your zapata.conf
accordingly.  I ended up setting my TX to -4.5 to cut out the choppiness.
 
Regards,
 
Mark.


  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 
Sent: Tuesday, March 14, 2006 12:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] echo problem + choppy sound


I have done this but I still get choppy sound and echo on some calls

thanks

Giovanni Miano [EMAIL PROTECTED] wrote: 

Of course,
Echo is 2 types: electric and ambiental.

If u gain rx o tx more than you need, its return in recive and gen echo

Try to decrase value, try to set 0 or .. in samecase -1 -2...


2006/3/13, sdgesa gaeharth [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] : 

Can you explain why? 


Giovanni Miano [EMAIL PROTECTED]  mailto:[EMAIL PROTECTED]  wrote:

rxgain=10.0
txgain=10.0

 

Maybe this is a problem



2006/3/13, sdgesa gaeharth   mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] : 

I still hear a slight echo of my voice when I talk with somone out the PSTN.
The voice on the other end sounds very choppy and a little distorted. When I
talk to other people within our office, the sound is perfect.Can some help?

We are all using:

Polycom 501 -- asterisk -- PSTN

zapata.conf:
[channels]
group = 1
language=en
context=incoming
signalling=fxs_ks 
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
musiconhold=default
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes 
transfer=yes
canpark=yes 
cancallforward=yes
callreturn=yes
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=10.0
txgain=10.0
channel = 1-4




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[Asterisk-Users] List Rules

[Bob McDowell]

Does anyone know if their are rules that this list is supposed to be
following?  It doesn't appear to be moderated, so I realize that such
rules would be self-enforced, but it still might be good to agree on
some.  Likewise, we could agree on none.  That works also.  Any
thoughts?

Some suggestions:

1) Be polite at all times.  Imagine your grandmother reads this list.

2) If you can't be productive (note that doesn't necessarily read
constructive), at least be brief.  And polite (as above).

3) Please post in (insert community approved format).  Likewise please
trim your posts (in 'X' preferred way).

4) (Etc)

N) If you are found in consistent breach of these rules, you will be
ignored until we can get you removed from the list.

Perhaps I'm the only one who would benefit from this, and if so I'll
promptly shut up and go back to work.


Thanks,

Bob McDowell



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Re: [Asterisk-Users] IAX choppy sound

[Tim Panton]

On 14 Mar 2006, at 16:44, Stojan Sljivic - GDS wrote:


Hi,

We have two Asterisk servers connected over IAX, with very limited 
bandwidth

256Kbs.
When we make calls between these two Asterisk servers the sound is very
choppy, no matter whether we use jitter buffer or not.

However, when we make calls using Skype, the sound is perfect.

Can anyone help us troubleshoot this IAX issue that we are 
experiencing?


Maybe if you tell us some more :-)
What codecs are you using?
Are you using Trunked IAX?
How many calls at a time?
What is the ping time between the systems?
Any error messages ?

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[Asterisk-Users] channel bridging

[JS]

Group:

Would it be possible to bridge two channels manually?

Scenario:

user1chanA-AsteriskchanBuser2

user3chanC-Asterisk

At this point, I send reINVITE to user2, and want to
bridge chanB with chanC and then tear down chanA.

My goal here is to make user2 talk with user3 instead
of talking to user1. Does anybody know an alternate way
of doing so?

Thanks,
Jim
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Re: [Asterisk-Users] Voip-Info

[Mojo with Horan Company, LLC]

I think this has been answered before.  Try searching in the list 
archives, or maybe the wiki :P


Douglas Garstang wrote:

Oh, I'm sorry. I must have missed the previous message where you specifically 
informed me not to use HTML.

-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 14, 2006 10:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voip-Info


On Tue, 2006-03-14 at 10:21 -0700, Douglas Garstang wrote:

Is it just me or is the voip-info web site down right now? Jeez that
web site is flaky.


Is it just me or was this message in HTML? Jeez some people never learn.


Dave Cotton [EMAIL PROTECTED]

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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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[Asterisk-Users] Voice volume using Monitor application

[Jeff Hoppe]

 I am using the Monitor() application (with
soxmix for combining the audios) and the voice connected to the phone network
is recorded at a lower volume then the voice connected directory to the Zap
analog phone card. How can I get both the audios to be at the same volume on
recording?



Thanks

Jeff






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Re: [Asterisk-Users] Attended Transfer - transfer timeout, how to change?

[Thomas Artner]

Am Tuesday 14 March 2006 18:38 schrieb Barry Flanagan:
 Hi,

 We are trying to use attended transfer with Asterisk 1.2.5, but when we
 do the transfer and dial the new number, it times out after 3 rings and
 then the callee is put back to the original agent.

 Where can I adjust the timeout which applies to the number we are
 transferring to? I have changed the extension for this number to timeout
 at 60 seconds, but that seems to make no difference.

try to adjust the parkingtime parameter in features.conf.

you are using the attended transfer feature.. 
ist it already possible to hang up before the other person lifts the handset 
without loosing the caller when you are doing an attendet transfer?

(person A takes an incoming call, person A would like to do an attended 
transfer to person B, person A hangs up the phone BEFORE person B takes the 
transfered call -- does the incoming call get lost?)

this was an issue in 1.2.4, I'd like to know whether its fixed in 1.2.5.

greets, 
Tom
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[Asterisk-Users] EICON Diva 4BRI

[Steve Kennedy]

Are there any step by step instrunctions on how to install drivers and I
guess bristuff for this card?

Just need to use it to handle voice on 2 BRI circuits (UK) then utilise
with Asterisk and some Digium cards handling POTS phones (and some VoIP
out the back).

It's the EICON card stuff and how to make it all work I'm finding
confusing?


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
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RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working

[Benjamin Lawetz]

Had this working also at some point, but had one killer problem... NAT
issues! Most of our clients are natted, and depending on the router, they
only allow traffic to return from the server that the traffic was sent to.
So the invites coming from other servers were being dropped.

But besides that worked like a charm.

Ben 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: March 14, 2006 12:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working

Yes, SIP realtime is working with multiple * servers all accessing the same
MySQL database, add a sip phone in the database and the phone can register
with any server without the need to configure any server, just add the phone
in the database, petty cool. 

JR

--

Message: 21
Date: Tue, 14 Mar 2006 10:18:06 -0500
From: Wai Wu [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

Now, I know what you guys been talking about. It is like DSN for sip phones,
not really clustering. I original thought that you guys want to setup some
thing that can fail over to a different sip server if the server running the
IVR dies.
 
 
 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, March 14, 2006 12:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working


Holy crap. You got SIP realtime working? I've tried it twice before and it
failed the same way twice. Do you have multiple Asterisk boxes accessing the
same sip info (ie phones) in the same table on the same database? Digium has
said numerous times this known not to work, although I cant' work out why as
it's just reading from a common table.


JR Richardson
Engineering for the Masses

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Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5

[Matt]

http://svn.digium.com/view/asterisk/team/oej/jitterbuffer-1.2/

On 3/14/06, Robert Webb [EMAIL PROTECTED] wrote:

 On Tue, 14 Mar 2006 14:32:02 +0100
   Olle E Johansson [EMAIL PROTECTED] wrote:
 
  14 mar 2006 kl. 13.35 skrev Matt:
 
  Right saw that.   But I'm trying to get away from using
 CVS-HEAD :)
  We all are. Every developer have switched from CVS to
 Subversion :-)
 
  This is not the development branch, but the release
 branch code,
  which we use to create the 1.2.x releases.
 
  The jitterbuffer itself is *not* release branch code,
 it's very much
  development. Please test it.
 
  The jitterbuffer branch is based on svn trunk (the
 same as the old  CVS HEAD)
  The jitterbuffer-1.2 branch is based on the 1.2 branch
 HEAD  (meaning latest 1.2 version code).
 
  /O

 Olle,

Pardon this dumb question please, but where are these
 test located. I looked under http://svn.digium.com and do
 not see them. I am not fluent in where everything is
 located and would like to do some testing on some of the
 other items such as the sip jitterbuffer. It will only be
 minimal but I would like to help where I can.

 Robert
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[Asterisk-Users] MWI Asterisk Realtime Architecture

[Ramin Nikaeen]

Hi Everyone,



I am using real time asterisk architecture and have placed
the following in sip.conf:



[general]

notifymimetype=text/plain

checkmwi=10

rtcachefriends=yes





but the MWI doesnt work?!



Can anyone give me any pointers as to what the problem could
be?



Thanks



ramin










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[Asterisk-Users] Outbound paging dialplan example?

[Patrick Friedel]

Due to changes at the office, I'm finally getting around to setting up 
an AA to deal with incoming calls.  One of the big changes is that we're 
dropping the old alphanumeric pager and will just send pages to our 
phones.  I've got the outbound greeting message working in a test 
context no problem right now, but I'm kind of stuck on how to capture a 
DTMF sequence from a user and doing anything with it.


Right now the pertinent DP features look like this:

exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,2
exten = s,5,Background(greeting)

exten = 1,1,Voicemail(u100) ; Press 1 to leave a message.

exten = 2,1,Voicemail(u6003) ; Press 2 to send an emergency page

exten = t,1,Dial(SIP/person,30,t) ; Ring my extension on timeout

Obviously extension 2 needs to be changed, right now it just leaves a 
message in my mailbox.  I'm figuring I'll add a new message that says 
Please enter your callback number, followed by the pound sign. and put 
that in as a Background() message.  The tricky bit that I can't figure 
out (without sample dialplans in voip-info) is how to capture the DTMF 
the caller provides and send it out via a System() call to an external 
application to page the oncall person.  As the oncall person will 
conceivably change on a regular basis, we can't just hand it out to 
customers, unfortunately/thankfully.  Thanks for any assistance!

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Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5

[Robert Webb]

On Tue, 14 Mar 2006 13:44:57 -0500
 Matt [EMAIL PROTECTED] wrote:

http://svn.digium.com/view/asterisk/team/oej/jitterbuffer-1.2/




Thank you I was looking directly under asterisk and 
not team. :-)


Robert
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RE: [Asterisk-Users] channel bridging

[Wai Wu]

Yes. Download the patch from here http://bugs.digium.com/view.php?id=5841


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
JSSent: Tuesday, March 14, 2006 1:15 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] channel 
bridging
Group:Would it be possible to bridge two channels 
manually?Scenario:user1chanA-AsteriskchanBuser2user3chanC-AsteriskAt 
this point, I send reINVITE to user2, and want tobridge chanB with chanC and 
then tear down chanA.My goal here is to make user2 talk with user3 
insteadof talking to user1. Does anybody know an alternate wayof doing 
so?Thanks,Jim
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RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working

[Wai Wu]

That is a show stopper. However, if your clients are in groups behind
their respected router, you might be able to give them a little linux
app such that this app can PERSONIFY the phones to send a packet to the
respected server.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Lawetz
Sent: Tuesday, March 14, 2006 1:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working

Had this working also at some point, but had one killer problem... NAT
issues! Most of our clients are natted, and depending on the router,
they only allow traffic to return from the server that the traffic was
sent to.
So the invites coming from other servers were being dropped.

But besides that worked like a charm.

Ben 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR
Richardson
Sent: March 14, 2006 12:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working

Yes, SIP realtime is working with multiple * servers all accessing the
same MySQL database, add a sip phone in the database and the phone can
register with any server without the need to configure any server, just
add the phone in the database, petty cool. 

JR

--

Message: 21
Date: Tue, 14 Mar 2006 10:18:06 -0500
From: Wai Wu [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

Now, I know what you guys been talking about. It is like DSN for sip
phones, not really clustering. I original thought that you guys want to
setup some thing that can fail over to a different sip server if the
server running the IVR dies.
 
 
 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, March 14, 2006 12:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Clustering NEW THREAD, Almost Working


Holy crap. You got SIP realtime working? I've tried it twice before and
it failed the same way twice. Do you have multiple Asterisk boxes
accessing the same sip info (ie phones) in the same table on the same
database? Digium has said numerous times this known not to work,
although I cant' work out why as it's just reading from a common table.


JR Richardson
Engineering for the Masses

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[Asterisk-Users] Realtime Extensions

[Douglas Garstang]

Does 
anyone know if realtime extensions allows extensions in the format 
callerid/extension yet? ie the extensions.conf file allows you to 
do:

5551212/1000 = exten ...

and it 
matches against extension 1000 when the caller id is 5551212. Last time I 
checked, realtime didn't support this yet.
That's 
a show stopper for us.



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[Asterisk-Users] Replicate 486 Sip Response Code

[Jon Weisman]

All,

How do I get Asterisk to return a 486 SIP response 
intentionally?

Thanks,
Jon
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Re: [Asterisk-Users] Outbound paging dialplan example?

[Doug Lytle]

Patrick Friedel wrote:
Obviously extension 2 needs to be changed, right now it just leaves a 
message in my mailbox.  I'm figuring I'll add a new message that says 
Please enter your callback number, followed by the pound sign. and 
put that in as a Background() message.  The tricky bit that I can't 
figure out (without sample dialplans in voip-info) is how to capture 
the DTMF the caller provides and send it out via a System() call to an 
external application to page the oncall


exten = 4852,1,Read(MEETMEPASS)|conf-getpin)
exten = 4852,n,Set(DB(conference/1000)=${MEETMEPASS})

Doug

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Re: [Asterisk-Users] Replicate 486 Sip Response Code

[Jon Weisman]

Hate to reply to my own post, but figured it out. 
Just have to setup the IP Phone to DND. 

Thanks,
Jon

  - Original Message - 
  From: 
  Jon Weisman 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, March 14, 2006 2:36 
  PM
  Subject: [Asterisk-Users] Replicate 486 
  Sip Response Code
  
  All,
  
  How do I get Asterisk to return a 486 SIP 
  response intentionally?
  
  Thanks,
  Jon
  
  

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[Asterisk-Users] ip telephony project

[JOSE MANUEL CORTES DAVID]

Hi

My name is Jose Manuel Cortes and im developing an 
IP telephony project, im going to interconnect a definity prologix PBX with an 
asterisk server (i still don't know what kind of cards i'll use digium, sangoma 
or voicetronix)trough a E1 connection in order to add ip telephony 
tothe university. Iwould like to know if there's a compatibility 
problem with this. I send you a diagram of the conection because the server is 
also a bridge for a trunk ( there's no more E1 ports in the PBX). Can you give 
me your opinion about the project and about what card shouldI 
buy?Can you help me find some info for the project. If I can help 
anyonejust let me know

Best regards 




Jose Manuel Cortes 
David
XSemestre Ingenieria 
Electronica
PONTIFICIA UNIVERSIDAD 
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[Asterisk-Users] Flash on Unicall Channel

[Paulo Scardine]

Hi all,

In Brazill, there is a trick to avoid collect calls: if you flash the 
line in the first 1000ms, Telco will drop any collect call for you.


Given the R3 signalling here, I have to use LibUnicall. Seems that there 
is no Flash command for unicall chanells, just for the Zap ones.


How can I flash the line?

I tried using Hungup command plus the h extension, but seems that 
Hungup is only a Goto(h|1), it does not trigger a drop call event on the 
channel.


How can I avoid collect calls then? I mean, how can I flash the Unicall 
channel?


Thanks in advance,
--
Paulo Scardine
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[Asterisk-Users] LCDPROC cient for Asterisk

[Mark Phillips]

I think I've asked this before and think that Matt had said something 
about this.


Is there an LCDproc client for Asterisk available and if so how can I 
get a copy please.


Thanks


--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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[Asterisk-Users] E911 from Remote Office via PRI

[Hugh L. Johnson]

Central business location has a PRI with a CLEC.  Remote offices access
the PRI for all voice traffic via VoIP.

How does one get the telco to report the address of a remote office to
the 911 call center when the call is made from that respective location?

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Re: [Asterisk-Users] Avaya IP Office 412

[Mark Phillips]

Do you have the right cable?

You need a cross-over T1 cable and NOT a cross-over ethernet cable that 
people commonly try. This should satify the electrical requirements and 
turn the lights green.


You're on your own with the rest.

I do have a question however; why are you now speaking SIP to the IP 
Office? Did you not buy that extra server?


Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


zgor wrote:

Hi!
First at all, sorry for my bad english ...
I m trying to connect an Avaya IP Office 412 to Asterisk using E1
I ve compiled/installed libpri - zaptel - asterisk correctly and now, 
im trying to get the link working.

I think, first step is to have green light on the  TE110P, isnt it?
I setup zaptel.conf:

span=1,1,0,css,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
defaultzone=es

So, i think: clock will be generated by Asterisk

But after making ztcfg -vv , i see that all channels are correctly 
setup, but running zttool, always i have RED Alarm


Any idea ?

Thanks you very much

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Re: [Asterisk-Users] E911 from Remote Office via PRI

[Peder @ NetworkOblivion]

Not to be a smarta**, but you have to ask them to do it.  We do the same 
thing and it works for us.  Depending on the CLEC, they may do it or 
they may say no.  If they say no, there isn't anything you can do about it.


Hugh L. Johnson wrote:

Central business location has a PRI with a CLEC.  Remote offices access
the PRI for all voice traffic via VoIP.

How does one get the telco to report the address of a remote office to
the 911 call center when the call is made from that respective location?

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http://www.networkoblivion.com

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