Re: [Asterisk-Users] accountcode not present in cdr

[Andres]

Max Clark wrote:


Hi all,

I have what I hope will be a simple problem. In my sip.conf I have 
defined the accountcode field (see below), and they system does not 
report any errors when I reload the configuration. However when I look 
at my cdr detail (either the csv on disk, or the mysql info) the 
accountcode that I have specified is missing. I have scoured the list 
and have seen a few postings on this with no solutions. What should I 
be looking at to debug this.


Thanks in advance,
Max


Setting the accountcode in sip.conf is totally unreliable.  It does not 
work in many cases.  Your best bet is to set it in a context via the 
command:

SetAccount([account]):  Set  the  channel account code for billing



I am running Asterisk 1.0.7 on CentOS 4.0:
Linux 2.6.9-11.EL #1 Wed Jun 8 16:59:52 CDT 2005 i686 i686 i386 GNU/Linux

[provider1]
accountcode=provider1
type=friend
host=10.1.1.1
dtmfmode=rfc2833
username=user
secret=12345
qualify=no
canreinvite=no
insecure=very
disallow=all
allow=ulaw
allow=gsm

[provider2]
type=peer
accountcode=provider2
secret=54321
username=user
host=10.1.1.10
dtmfmode=rfc2833





--
Andres
Network Admin
http://www.telesip.net


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[Asterisk-Users] asterisk and radius?

[maka]

Hi all,

I have been looking at some billing solutions for asterisk. I saw
there is Trabas VoIP Billing which apparently is working through
radius cdr records, and also astPP which was recently released, and
CDRTool.

Has anyone been able to succeffully use radius with asterisk for CDR records? 

I tried app_radius with freerdius according to the wiki docs, but the
Radius(CPP) keeps playing a prompt for pin and password - can't that
be bypassed in some way - I don't want each call to be interactively
authenticated on the asterisk since the client has already registered
with SER, I just need the CDR.

Or does that mean that I have just misconfigured freeradius and radius.conf?

If anyone has a more-extencive document on configuring asterisk with
freeradius, please post an advice.

Thank you.
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Re: [Asterisk-Users] ee1000 Ethernet in Dell 1850

[Andres]

Andres wrote:






Digium's site now lists the Dell 1850 as a potential problem server, 
as it uses the intel ee1000 Ethernet chipset (as do a majority of 
servers in the market!).


To my knowledge, ALL dell servers with Gigabit interfaces now use 
the same chipset. Does this mean the Digium cards can't be used in 
Dell servers unless you disable the onboard ethernet?


I don't want to disable the onboard interface, as I use the IPMI 
management facility for lights-out management. I have a 2850 that 
doesn't have any audio problems (the reason that I contacted Digium 
in the first place), so I'm wondering if Digium are simply guessing 
at problems.


Does anyone know anything specific about the supposed 
incompatibilities with the ee1000 kernel module?



I am not sure where you got that chipset reference but all our 
PowerEdge 1850s come with:

Ethernet controller: Intel Corp. 82541GI/PI Gigabit Ethernet Controller

...and they work fine with the TE410.


I finally got to confirm the e1000 reference on our systems:

[EMAIL PROTECTED] net]# cat /proc/ioports

0cf8-0cff : PCI conf1
bc00-bcff : ATI Technologies Inc Radeon RV100 QY [Radeon 7000/VE]
c000-dfff : PCI Bus #05
 c000-cfff : PCI Bus #07
   ccc0-ccff : PCI device 8086:1076 (Intel Corp.)
 ccc0-ccff : e1000
 d000-dfff : PCI Bus #06
   dcc0-dcff : PCI device 8086:1076 (Intel Corp.)
 dcc0-dcff : e1000

But as I said, the TE410 works perfectly under RH ES3.0







--
Andres
Network Admin
http://www.telesip.net


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RE: [Asterisk-Users] Asterisk does not function without a DNS ser ver

[Guido Hecken]

 We have our Asterisk server running smoothly with a SIP BRI gateway
 for inbound calls. However if the Internet connection goes down and a
 DNS server becomes unreachable Asterisk basically does not function.
 By this I mean it does not answer call coming in from the gateway
 (which is on the local LAN) and you can't even reload it - just hangs
 there. If I change the DNS setting in resolv.conf to something else
 which is reachable all is well again.
 
 I have tried setting srvlookup=no in sip.conf but it made no difference.
 
 Does anyone know how I to make Asterisk continue working for local LAN
 users/gateways when a DNS server is not reachable?

Try to use bind on the * Machine and configure it as a caching only
nameserver.

Hope, this helps

Regards,

Guido Hecken
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[Asterisk-Users] Re: $0-per-month (pay as you go) provider with T.38?

[Adam Megacz]

Joshua Colp [EMAIL PROTECTED] writes:
 Make sure you're not using asterisk or you will have no T.38
 support, not even passthrough.

Isn't this sort of like saying make sure you're not using Apache, or
you will have no SMTP support?

I call out to t38modem from an AGI script.  You know, that whole
extensibility thing.

  - a


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Re: [Asterisk-Users] Compilation Problem with asterisk-addons

[Nico Giefing]

- Original Message - 
From: Juan Luis Moyano [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 6:09 AM
Subject: Re: [Asterisk-Users] Compilation Problem with asterisk-addons


 On Lun, 20 de Junio de 2005, 6:49 pm, Nico Giefing dijo:
  Hello, i have a little Problem with compiling asterisk-addons
 
 
  the failure is:
 
  app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4
  arguments, but only 3 given
  app_addon_sql_mysql.c: In function `del_identifier':
  app_addon_sql_mysql.c:164: error: ÀSR_LIST_REMOVE' undecalred (first use
  in this function)
  app_addon_sql_mysql.c:164: error: (Each undeclared identifier is
reported
  only once
  app_addon_sql_mysql.c:164: error: for each function it appears in.)
 
 
  Does anybody know anything about this problem?
 
 
  Thank you for your help.
 
 
  Nico
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 Nico, I'm having the same issue while compiling asterisk-addons. Here I
 post the error I get:

 app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4
 arguments, but only 3 given
 app_addon_sql_mysql.c: In function `del_identifier':
 app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use
 in this function)
 app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported
 only once
 app_addon_sql_mysql.c:164: error: for each function it appears in.)
 make: *** [app_addon_sql_mysql.o] Error 1

 Also it's important to mention that I'm running asterisk-1.0.7 compiled
 from the ebuild on a Gentoo (kernel 2.6), also I've merged latest mysql,
 perl and DBD-mysql. I don't know what is the best way to compile
 asterisk-addons on a gentoo system so if someone had accomplished this,
 please let me know. Thanks in advance.

 -- 
 Juan Luis Moyano
 [EMAIL PROTECTED]

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Hi, my configuration is debian stable (3.1) with 2.6 kernel and the cvs
version of asterisk and asterisk-addons.

does anybody know a solution for this problem?

Nico


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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 131

[Nguyen Trung Tin]

Hi All
I wan to get DTMF while voicemail recording sounds. DTMF received save to contents field of mail attach with wave sounds file.
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RE: [Asterisk-Users] Panasonic KX-TD1232

[Peter Svensson]

On Mon, 20 Jun 2005, Dan Morin wrote:

 Can you let me know what hardware you are using and how the two systems
 are configured to work together?  Thanks in advance.

We have an E1 PRI card in the KX-TD1232 and a TE405P in the Asterisk box. 
The Asterisk box sits between the pstn and the KX-TD1232. 

 Can anyone confirm that dialing 8 + the Trunk Group number will select a
 CO line in that trunk?  Thanks in advance.

The exact digits to dial to request a specific trunk group can be changed, 
but it defaults to 8. 

Peter

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Re: [Asterisk-Users] Re: Libtiff 3.5.7 - recommended version for spandsp

[Marco Parmeggiani]

Roger Schreiter ha scritto:

Marco Parmeggiani wrote:
  ...


i had no problems receiving faxes with version 3.7.2.
on the other hand i have big problems in sending multipage faxes. only 




Hi,

where did you get that version?
On libtiff.org, 3.6.1 is the most recent one.



you're pointing to the wrong page:
http://www.remotesensing.org/libtiff/
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Re: [Asterisk-Users] webvmail debian package

[Tzafrir Cohen]

Hi

On Mon, Jun 20, 2005 at 09:25:13AM +0200, sylvain garcia wrote:
 Hi,
 
 I wouldlike use webvmail on my asterisk, I use debain Sarge with
 asterisk 1.0.7 package.
 I have installed package asterisk-web-vmail but when i go to
 http://MyAsteriskBOx, i have a page of presentation of Apache.

The list of files in the package:

http://packages.debian.org/cgi-bin/search_contents.pl?searchmode=filelistword=asterisk-web-vmailversion=stablearch=all

/usr/lib/cgi-bin is Debian's /cgi-bin . Thus the script is by-default at
http://hostname/cgi-bin/asterisk/vmail.cgi

Also note that you need to allow the script to read/write
/etc/asterisk/voicemail.conf and /var/spool/asterisk/voicemail/ .
This can basically be done in two ways:

1. add the user www-data (apache) to the group asterisk 
   - Every apache script has full access to asterisk
   - and if you still run asterisk as root, this is a potential root
 exploit

2. This is a separate cgi script: use suexec to run it as the asterisk
   user. However I must admit I found no straight-forward way of doing
   it for that script alone.

However in Rapid the vmail script is currently disabled: I wanted to try
to adapt it to our slightly different voicemail configuration and
noticed that the script has quite a bit of sphagetty in it. . I didn't
want to start modifying perl code that does not use strict and didn't
find the time to rewrite it to use strict.

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

[Tzafrir Cohen]

On Mon, Jun 20, 2005 at 12:46:27AM -0700, Syed Akbar wrote:
 Does anyone have any comments about using Debian stable release Vs Fedora
 for running Asterisk?

Quite a flame bait, so I'd like to make it even more so: With Debian
stable you have two completely different options:

- Use the official Debian packages

- Build your own packages, just like you can on Fedora.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation

[David Wilson]

Hi guys,

I'm running a Sirrix PCI4S0 quad BRI card in a 
box with Asterisk CVS-HEAD (20050614).
Something I've come across is that with 
'echocancel = yes' in /etc/asterisk/sirrix.conf the echo is unbearably loud - so 
loud in fact that the echo distorts.

To remedy this I've set 'echocancel = no' and 
disabled the echo cancellation. With the echo cancellation disabled there is 
still an echo but it is much softer.

Any ideas on how I can turn on the echo 
cancellationagain without having the veryloud echo back 
?
Is there some way I could perhaps drop the TX 
volume out of the Sirrix card ? Perhaps this would help ?

Thanks in advance.

Kindest regardsDavid Wilson___D 
c D a t aTel +27 33 342 7003Fax +27 33 345 4155Cell +27 82 
4147413http://www.dcdata.co.za[EMAIL PROTECTED]Powered by Linux, 
driven by passion ! ___

"Computers are not intelligent. They only think they 
are."
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Re: [Asterisk-Users] ee1000 Ethernet in Dell 1850

[Zoa]

Hmm, i dont think thats the reason they dont recommend the dell server.
The problems with the ee1000 kernel module are easily resolved, compile
the module into the kernel.

Zoa,

Andres wrote:






Digium's site now lists the Dell 1850 as a potential problem server,
as it uses the intel ee1000 Ethernet chipset (as do a majority of
servers in the market!).

To my knowledge, ALL dell servers with Gigabit interfaces now use
the same chipset. Does this mean the Digium cards can't be used in
Dell servers unless you disable the onboard ethernet?

I don't want to disable the onboard interface, as I use the IPMI
management facility for lights-out management. I have a 2850 that
doesn't have any audio problems (the reason that I contacted Digium
in the first place), so I'm wondering if Digium are simply guessing
at problems.

Does anyone know anything specific about the supposed
incompatibilities with the ee1000 kernel module?




I am not sure where you got that chipset reference but all our
PowerEdge 1850s come with:
Ethernet controller: Intel Corp. 82541GI/PI Gigabit Ethernet Controller

...and they work fine with the TE410.



There seems to be an ever-growing list of situations where you can't
use the Digium cards. This is a concern to me.
___











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Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation

[altus]

On the subject of this
in you /var/log/messages do you get errors like this

Jun 21 09:38:45 pbxct sshd(pam_unix)[1993]: session opened for user root by 
(uid=0)
Jun 21 09:39:22 pbxct kernel: 00: Unsetting old PCM master PFIC 0
Jun 21 09:39:22 pbxct kernel: 00: Setting PCM master to PFIC 0 IPAC3
Jun 21 09:39:22 pbxct kernel: Slip detected on IPAC3
Jun 21 09:40:02 pbxct kernel: sirrix ipac (3): ipac_handle_interrupt_icd: 
error, RSTAD = 0x1e not ok!
Jun 21 09:40:07 pbxct kernel: Slip detected on IPAC3

email this guy,he wrote a patch to bring down the volume
[EMAIL PROTECTED]

On Tue, 2005-06-21 at 09:37 +0200, David Wilson wrote:
 Hi guys,
  
 I'm running a Sirrix PCI4S0 quad BRI card in a box with Asterisk CVS-
 HEAD (20050614).
 Something I've come across is that with 'echocancel = yes'
 in /etc/asterisk/sirrix.conf the echo is unbearably loud - so loud in
 fact that the echo distorts.
  
 To remedy this I've set 'echocancel = no' and disabled the echo
 cancellation. With the echo cancellation disabled there is still an
 echo but it is much softer.
  
 Any ideas on how I can turn on the echo cancellation again without
 having the very loud echo back ?
 Is there some way I could perhaps drop the TX volume out of the Sirrix
 card ? Perhaps this would help ?
  
 Thanks in advance.
  
 
 Kindest regards
 David Wilson
 ___
 D c D a t a
 Tel +27 33 342 7003
 Fax +27 33 345 4155
 Cell +27 82 4147413
 http://www.dcdata.co.za
 [EMAIL PROTECTED]
 Powered by Linux, driven by passion ! 
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 Computers are not intelligent. They only think they are.
 
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RE: [Asterisk-Users] help for making several calls at the same time..

[Erdem HAKİ]

Hi 



I surmounted the problem by myself :), when
i add user who has 12 digits number like 902121112233 ,everything
works fine. 



Erdem HAKI  [EMAIL PROTECTED]











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKİ
Sent: Monday, June 20, 2005 9:57
AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] help for
making several calls at the same time..





Hi,



I have installed latest stable version of Asterisk. I
registered 1 Quintum Gateway and 5 eyeBeam SoftPhone. But I can make just one
call at the same time, if i try to make calls from 2 softphones to anotherone, second
caller listens  the person have extension  is on the phone
 . So we couldnt make two or more calls at the same time for
a SoftPhone. What should we do to make several calls at the same time?



Thanks for your interest. 



Erdem HAKI  [EMAIL PROTECTED]






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Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation

[David Wilson]

Hi Altus,

Thanks for your reply.

Yes I do get those errors. Any ideas what causes them ?


email this guy,he wrote a patch to bring down the volume

Thanks I've been chatting with Steve already.

For some reason the patch does not seem to be working with my newer version 
of the Sirrix driverseither that or I'm doing something wrong. :)
Did you have to modify the patch in any way or did you just apply it 'as is' 
?


I will keep in touch to let you know the outcome.

Many thanks.

Kindest regards
David Wilson
___
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Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
___

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- Original Message - 
From: altus [EMAIL PROTECTED]

To: asterisk asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 9:52 AM
Subject: Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation



On the subject of this
in you /var/log/messages do you get errors like this

Jun 21 09:38:45 pbxct sshd(pam_unix)[1993]: session opened for user root 
by (uid=0)

Jun 21 09:39:22 pbxct kernel: 00: Unsetting old PCM master PFIC 0
Jun 21 09:39:22 pbxct kernel: 00: Setting PCM master to PFIC 0 IPAC3
Jun 21 09:39:22 pbxct kernel: Slip detected on IPAC3
Jun 21 09:40:02 pbxct kernel: sirrix ipac (3): ipac_handle_interrupt_icd: 
error, RSTAD = 0x1e not ok!

Jun 21 09:40:07 pbxct kernel: Slip detected on IPAC3

email this guy,he wrote a patch to bring down the volume
[EMAIL PROTECTED]

On Tue, 2005-06-21 at 09:37 +0200, David Wilson wrote:

Hi guys,

I'm running a Sirrix PCI4S0 quad BRI card in a box with Asterisk CVS-
HEAD (20050614).
Something I've come across is that with 'echocancel = yes'
in /etc/asterisk/sirrix.conf the echo is unbearably loud - so loud in
fact that the echo distorts.

To remedy this I've set 'echocancel = no' and disabled the echo
cancellation. With the echo cancellation disabled there is still an
echo but it is much softer.

Any ideas on how I can turn on the echo cancellation again without
having the very loud echo back ?
Is there some way I could perhaps drop the TX volume out of the Sirrix
card ? Perhaps this would help ?

Thanks in advance.


Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
___

Computers are not intelligent. They only think they are.

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回复： [Asterisk-Users] storing CDR record s in a MySQL database

[Gary Li]

Hi,
Using what database name is not the matter.
You should configure the dbname and user password in the /etc/asterisk/cdr_mysql.conf.


Joseph [EMAIL PROTECTED] 写道：
I'm trying to configure CDR records to store them in MySQL database butthose instruction are not very clear from:http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysqlI've MySQL up and running I'm just not sure what database to create andas the configuration Sample cdr_mysql.conf is not consistent frominstruction on: Create the databaseDo I create database name "cdr" or "asteriskcdrdb"?Whatever name I use I get an error creating the tables.-- #Joseph___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
Best Regards,
Gary Li
		DO YOU YAHOO!? 
 
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Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation

[altus]

No,sometimes i get a watery sound,like when you speak under water
I turned echo off,do you have the latest driver.A new version came out
on the 16th
Altus
On Tue, 2005-06-21 at 10:17 +0200, David Wilson wrote:
 Hi Altus,
 
 Thanks for your reply.
 
 Yes I do get those errors. Any ideas what causes them ?
 
 email this guy,he wrote a patch to bring down the volume
 Thanks I've been chatting with Steve already.
 
 For some reason the patch does not seem to be working with my newer version 
 of the Sirrix driverseither that or I'm doing something wrong. :)
 Did you have to modify the patch in any way or did you just apply it 'as is' 
 ?
 
 I will keep in touch to let you know the outcome.
 
 Many thanks.
 
 Kindest regards
 David Wilson
 ___
 D c D a t a
 Tel +27 33 342 7003
 Fax +27 33 345 4155
 Cell +27 82 4147413
 http://www.dcdata.co.za
 [EMAIL PROTECTED]
 Powered by Linux, driven by passion !
 ___
 
 Computers are not intelligent. They only think they are.
 
 - Original Message - 
 From: altus [EMAIL PROTECTED]
 To: asterisk asterisk-users@lists.digium.com
 Sent: Tuesday, June 21, 2005 9:52 AM
 Subject: Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation
 
 
  On the subject of this
  in you /var/log/messages do you get errors like this
 
  Jun 21 09:38:45 pbxct sshd(pam_unix)[1993]: session opened for user root 
  by (uid=0)
  Jun 21 09:39:22 pbxct kernel: 00: Unsetting old PCM master PFIC 0
  Jun 21 09:39:22 pbxct kernel: 00: Setting PCM master to PFIC 0 IPAC3
  Jun 21 09:39:22 pbxct kernel: Slip detected on IPAC3
  Jun 21 09:40:02 pbxct kernel: sirrix ipac (3): ipac_handle_interrupt_icd: 
  error, RSTAD = 0x1e not ok!
  Jun 21 09:40:07 pbxct kernel: Slip detected on IPAC3
 
  email this guy,he wrote a patch to bring down the volume
  [EMAIL PROTECTED]
 
  On Tue, 2005-06-21 at 09:37 +0200, David Wilson wrote:
  Hi guys,
 
  I'm running a Sirrix PCI4S0 quad BRI card in a box with Asterisk CVS-
  HEAD (20050614).
  Something I've come across is that with 'echocancel = yes'
  in /etc/asterisk/sirrix.conf the echo is unbearably loud - so loud in
  fact that the echo distorts.
 
  To remedy this I've set 'echocancel = no' and disabled the echo
  cancellation. With the echo cancellation disabled there is still an
  echo but it is much softer.
 
  Any ideas on how I can turn on the echo cancellation again without
  having the very loud echo back ?
  Is there some way I could perhaps drop the TX volume out of the Sirrix
  card ? Perhaps this would help ?
 
  Thanks in advance.
 
 
  Kindest regards
  David Wilson
  ___
  D c D a t a
  Tel +27 33 342 7003
  Fax +27 33 345 4155
  Cell +27 82 4147413
  http://www.dcdata.co.za
  [EMAIL PROTECTED]
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  ___
 
  Computers are not intelligent. They only think they are.
 
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Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation

[David Wilson]

I'm using the ones from 10/06. I haven't put in the 16/06 ones yet. Have you 
tried them ?



I turned echo off

Is your echo cancellation switched off  (echocancel = no) ?

Yea, I also get a 'watery' sound now and again.

With my echocancel set to 'no' everything works fairly well except that 
there's a slight echo, much softer than the echo I get when I have 
'echocancel = yes' with or without the patch applied.


Hows JHB today ? :)

Kindest regards
David Wilson
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Cell +27 82 4147413
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Computers are not intelligent. They only think they are.

- Original Message - 
From: altus [EMAIL PROTECTED]

To: David Wilson [EMAIL PROTECTED]
Cc: asterisk asterisk-users@lists.digium.com; Steve Davies 
[EMAIL PROTECTED]

Sent: Tuesday, June 21, 2005 10:23 AM
Subject: Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation



No,sometimes i get a watery sound,like when you speak under water
I turned echo off,do you have the latest driver.A new version came out
on the 16th
Altus
On Tue, 2005-06-21 at 10:17 +0200, David Wilson wrote:

Hi Altus,

Thanks for your reply.

Yes I do get those errors. Any ideas what causes them ?

email this guy,he wrote a patch to bring down the volume
Thanks I've been chatting with Steve already.

For some reason the patch does not seem to be working with my newer 
version

of the Sirrix driverseither that or I'm doing something wrong. :)
Did you have to modify the patch in any way or did you just apply it 'as 
is'

?

I will keep in touch to let you know the outcome.

Many thanks.

Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
___

Computers are not intelligent. They only think they are.

- Original Message - 
From: altus [EMAIL PROTECTED]

To: asterisk asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 9:52 AM
Subject: Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo 
cancellation



 On the subject of this
 in you /var/log/messages do you get errors like this

 Jun 21 09:38:45 pbxct sshd(pam_unix)[1993]: session opened for user 
 root

 by (uid=0)
 Jun 21 09:39:22 pbxct kernel: 00: Unsetting old PCM master PFIC 0
 Jun 21 09:39:22 pbxct kernel: 00: Setting PCM master to PFIC 0 IPAC3
 Jun 21 09:39:22 pbxct kernel: Slip detected on IPAC3
 Jun 21 09:40:02 pbxct kernel: sirrix ipac (3): 
 ipac_handle_interrupt_icd:

 error, RSTAD = 0x1e not ok!
 Jun 21 09:40:07 pbxct kernel: Slip detected on IPAC3

 email this guy,he wrote a patch to bring down the volume
 [EMAIL PROTECTED]

 On Tue, 2005-06-21 at 09:37 +0200, David Wilson wrote:
 Hi guys,

 I'm running a Sirrix PCI4S0 quad BRI card in a box with Asterisk CVS-
 HEAD (20050614).
 Something I've come across is that with 'echocancel = yes'
 in /etc/asterisk/sirrix.conf the echo is unbearably loud - so loud in
 fact that the echo distorts.

 To remedy this I've set 'echocancel = no' and disabled the echo
 cancellation. With the echo cancellation disabled there is still an
 echo but it is much softer.

 Any ideas on how I can turn on the echo cancellation again without
 having the very loud echo back ?
 Is there some way I could perhaps drop the TX volume out of the Sirrix
 card ? Perhaps this would help ?

 Thanks in advance.


 Kindest regards
 David Wilson
 ___
 D c D a t a
 Tel +27 33 342 7003
 Fax +27 33 345 4155
 Cell +27 82 4147413
 http://www.dcdata.co.za
 [EMAIL PROTECTED]
 Powered by Linux, driven by passion !
 ___

 Computers are not intelligent. They only think they are.

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[Asterisk-Users] ipswitchboard

[altus]

Good day all
Im trying to download ipswitchboard but the webpage does not seem to
work?
Can someone maybe put it somewhere,and the .NET thing you must install
with it,please
Or is there a different link to http://ipswitchboard.thorben.dk/
Please 
Thanks
ALtus

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[Asterisk-Users] [sourceforge.net abuse] New JAVA application server for Asterisk - OrderlyCalls

[Adam Megacz]

Matt,

Sourceforge.net is exclusively for hosting software whose licensing
terms meet the OSI's definition of Open Source:

  http://opensource.org/docs/definition.php

Your licensing terms include the following, which is not compliant
with the OSI definition:

  Usage Restrictions

  In addition to the restrictions of the LGPL, the following
  restrictions apply: ...  OrderlyCalls may not be used to provide or
  augment call queuing without the prior written permission of Orderly
  Software.

While I understand your motivation and empathize with the plight of
open-source business, unfortunately you must either:

  a) remove this restriction

- or -

  b) remove your project from sourceforge.net

Please take action soon so that this matter does not need to be
escalated to the sourceforge.net admins.

  - a

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Re: [Asterisk-Users] OH323 with g723

[Bashir Ullah - www.Lamsre.Com]

Hi

I did install and i bought g729 from digium. and my g723 works fine also
with quintum but i have now another problem, i found robotic sound. and
delay of my outgoing voice. incoming voice is fine.


bashir




- Original Message - 
From: Erdem HAKİ [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, June 20, 2005 1:11 AM
Subject: RE: [Asterisk-Users] OH323 with g723


 Hi,

 Visit http://aussievoip.com.au/wiki-G723-1-Install you'll find how to
 install g723, but first you have to install g729
 http://aussievoip.com.au/wiki-G729-Install

 I have tested it with Quintum, it works

 Enjoy :)

 Erdem HAKI - [EMAIL PROTECTED]


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Bashir
Ullah -
 www.Lamsre.Com
 Sent: Monday, June 20, 2005 11:08 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] OH323 with g723

 hi

 is there anybody using g723 with oh323 and sending call by asterisk. if so
 please let me know how i can use this same, i need to call quintum by g723
.

 Thanks
 Bashir

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Re: [Asterisk-Users] chan_h323 vs chan_oh323 chan_ooh323

[Bashir Ullah - www.Lamsre.Com]

Please post ur installation script for chan_h323
 



- Original Message - 
From: Atif Rasheed [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, June 20, 2005 7:21 AM
Subject: [Asterisk-Users] chan_h323 vs chan_oh323  chan_ooh323


 hello there,
 can somebody please comment which one of these channel drivers will give 
 best output doing g729|g723 pass-thru. only pass-thru is needed no 
 transcoding.
 please share your experience. if somebody has some figures (simultanous 
 calls using a certain channel driver) it will be apericiated. I have 
 installed chan_h323 (by McNamara) and its working fine with me. I just 
 want  to know if I run this driver on a Dual-Xeon machine. can it handle 
 500 or  500 simultanous calls in pass-thru mode.
 
 Regards,
 --
 Atif
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RE: [Asterisk-Users] voicemail system

[harry gaillac]

Hello,

In fact i wish to add users table in ser DB. 
What's voicemessages table?

How can i configure voicemail conf according to
extconf
and voicemail messages stored in db.

readme.extconfig say voicemail conf is stored in db.
readme.odbcstorage say voicemail message are stored in
other table

What about ast_data ?

regards

Harry

--- Orlando Guitián [EMAIL PROTECTED] a écrit :

 4. Configure database
 Create a database (e.g. 'asterisk_vm')  a user
 which can access it (needs 
 to have write access for password changes from
 inside VM).
 
 For mysql table has to be called 'users' (hardcoded
 in .h file)
 
 CREATE TABLE users (
context char(79) DEFAULT '' NOT NULL,
mailbox char(79) DEFAULT '' NOT NULL,
password char(79) DEFAULT '' NOT NULL,
fullname char(79) DEFAULT '' NOT NULL,
email char(79) DEFAULT '' NOT NULL,
pager char(79) DEFAULT '' NOT NULL,
options char(159) DEFAULT '' NOT NULL,
stamp timestamp,
PRIMARY KEY (context,mailbox)
 );
 
 For postgres table has to be called 'voicemail'
 
 CREATE TABLE voicemail (
context varchar(79) DEFAULT '' NOT NULL,
mailbox varchar(79) DEFAULT '' NOT NULL,
password varchar(79) DEFAULT '' NOT NULL,
fullname varchar(79) DEFAULT '' NOT NULL,
email varchar(79) DEFAULT '' NOT NULL,
pager varchar(79) DEFAULT '' NOT NULL,
options varchar(159) DEFAULT '' NOT NULL,
stamp timestamp,
PRIMARY KEY (context,mailbox)
 );
 
 Note that context refers to the Mailbox context, not
 extension context)
 Note that the password is stored in plain text
 
 
 From: harry gaillac [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List -
 Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 To: Asterisk-Users@lists.digium.com
 Subject: [Asterisk-Users] voicemail system
 Date: Tue, 21 Jun 2005 00:51:48 +0200 (CEST)
 
 Hello,
 
 I wish to use asterisk as a voicemail server with
 ser
 .
 
 I want to use asterisk external configuration
 toHello,
 
 I wish to use asterisk as a voicemail server with
 ser
 .
 
 I want to use asterisk external configuration to
 manage users and storing voicemail messages
 according
 to ser database.
 
 Where can i find the schema of the SQL DB for
 voicemail accounts .
 for example in extconfig ; file.conf =
 driver,database[,table]
 ;voicemail = odbc,asterisk
 
 where is the schema of asterisk database ?
 
 which script retrieve values in db for
 voicemail.conf
 ?
 
 Regards
 
 Harry






___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com

README.extconfig
Description: 4218501336-README.extconfig


README.odbcstorage
Description: 1455846133-README.odbcstorage
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Re: [Asterisk-Users] Asterisk does not function without a DNS server

[dsr]

On Tue, Jun 21, 2005 at 07:56:29AM +1000, Eric Bishop wrote:
 We have our Asterisk server running smoothly with a SIP BRI gateway
 for inbound calls. However if the Internet connection goes down and a
 DNS server becomes unreachable Asterisk basically does not function.
 By this I mean it does not answer call coming in from the gateway
 (which is on the local LAN) and you can't even reload it - just hangs
 there. If I change the DNS setting in resolv.conf to something else
 which is reachable all is well again.
 
 I have tried setting srvlookup=no in sip.conf but it made no difference.
 
 Does anyone know how I to make Asterisk continue working for local LAN
 users/gateways when a DNS server is not reachable?

How about running a DNS caching server on the same machine as
Asterisk?

-dsr-
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Re: [Asterisk-Users] Asterisk does not function without a DNS ser ver

[Eric Bishop]

I'd really rather not run a DNS server if I don't have to. Surely ther
must be a way to tell Asterisk not to rely on DNS?



On 6/21/05, Guido Hecken [EMAIL PROTECTED] wrote:
  We have our Asterisk server running smoothly with a SIP BRI gateway
  for inbound calls. However if the Internet connection goes down and a
  DNS server becomes unreachable Asterisk basically does not function.
  By this I mean it does not answer call coming in from the gateway
  (which is on the local LAN) and you can't even reload it - just hangs
  there. If I change the DNS setting in resolv.conf to something else
  which is reachable all is well again.
 
  I have tried setting srvlookup=no in sip.conf but it made no difference.
 
  Does anyone know how I to make Asterisk continue working for local LAN
  users/gateways when a DNS server is not reachable?
 
 Try to use bind on the * Machine and configure it as a caching only
 nameserver.
 
 Hope, this helps
 
 Regards,
 
 Guido Hecken

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RE: [Asterisk-Users] Looking for PRI Outbound Caller ID Configuration

[Rich Adamson]

 That didn't seem to work either.  Any other ideas?
 
 Thanks,
 
 Shaun
 
 -Original Message-
  I'm having trouble setting the outbound caller ID on calls I make from my
  PRI trunk group.  The PRIs are served out of a 5ESS.  Telco has set the
 PRIs
  up for user provided caller id information, so I believe I just don't have
  it set up right in my dialplan or something.  I can't seem to find an
  example of setting the outbound caller ID specifically for a 5ESS.  Does
  anyone have an example configuration that they have used with a 5ESS
 switch?
  Below is the my configuration from Zapata.conf and a sample extension I've
  tried to use to connect a call with new caller ID information provided by
 my
  PBX.  Any insight is most appreciated.
  
  [channels]
  priindication = outofband
  usecallerid=yes
  cidsignalling=bell
  hidecallerid=no
  callwaiting=no
  usecallingpres=yes
  callwaitingcallerid=no
  threewaycalling=yes
  transfer=yes
  cancallforward=yes
  callreturn=yes
  echocancel=yes
  echocancelwhenbridged=no
  rxgain=0.0
  txgain=0.0
  group=1
  callgroup=1
  pickupgroup=1
  immediate=no
  switchtype=5ess
  context=main
  signalling=pri_cpe
  group=1
  channel = 1-23
  channel = 25-47
  
  exten = 1234,1,Wait,1
  exten = 1234,2,Answer
  exten = 1234,3,SetCallerPres(allowed_passed_screen)
  exten = 1234,4,SetCIDNum(8881234567)
  exten = 1234,5,Dial(Zap/g1/18887654321,,,)
  exten = 1234,6,Hangup
  
 
 Try something like this...
 exten = _1NX,1,SetCallerID(8881234567|a)
 exten = _1NX,2,SetCIDName(MyName|a)
 exten = _1NX,3,Dial(ZAP/g1/${EXTEN})

If the above doesn't seem to work, the next step that I'd take is to
have the central office tech's trace a call for you. They _can_
do that. 

The last pri that I implemented had similar problems, and it only
took a few minutes with the right people on the phone to resolve
it. Getting to the right person in the CO tends to be a problem
in some cases. (In my case, they swore up and down things were
configured correctly on their end. But after a couple of traces
they found one of their switches was dropping callerid info.)


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[Asterisk-Users] Problems with wew FXO modules for TDM400P

[robert.brown01]

Has anyone else had problems with these new modules?



I was using 2 x X100P, but wanted to try something more
stable, so got 2 brand new FXO modules for my TDM400P. I
have to analogue lines from different providers (BT and Telewest), where Module
1 (BT) is not detecting pickup as the incoming call is still ringing out, but
has been accepted by an extension. Module 2 is connecting and a call is
generated, however, the incoming call is not detecting when the extension hangs
up. 



Asterisk version 1.07 stable, ([EMAIL PROTECTED] 1.1)



I was made aware that there had been a few delays in these modules
being released, so any help or confirmation that this is a know problem would
be greatly appreciated.





Robert Brown



UK










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Re: [Asterisk-Users] Unable to reconfigure channel

[Rich Adamson]

The reload command does not re-implement every single parameter change,
therefore in many cases one has to stop  restart asterisk.

 
 I have a problem with the cvs head zaptel library: I cannot update my 
 zapata.conf into 
asterisk when I issue the reload command from the CLI
 prompt; only when I stop and restart the asterisk service.
 
  
 
 The system sends the following message:
 
  
 
 Jun 20 09:49:14 WARNING[3754]: chan_zap.c:10364 setup_zap: 
 Ignoring switchtype
 
 Jun 20 09:49:14 WARNING[3754]: chan_zap.c:10364 setup_zap: Ignoring signalling
 
 -- Reconfigured channel 1, FXS Kewlstart signalling
 
 -- Reconfigured channel 2, FXS Kewlstart signalling
 
 -- Reconfigured channel 3, FXS Kewlstart signalling
 
 Jun 20 09:49:14 ERROR[3754]: chan_zap.c:9827 setup_zap: Unable to reconfigure 
 channel '4'
 
 Jun 20 09:49:14 WARNING[3754]: chan_zap.c:10567 reload: Reload of chan_zap.so 
 is unsuccessful!
 
  
 
 I have the simplest configuration:
 
  
 
 [trunkgroups]
 
 [channels]
 
 language=en
 
 context=default
 
 switchtype=national
 

 
 signalling=fxs_ks
 
 channel = 1
 
 channel = 2
 
 channel = 3
 
  
 
 I was using the cvs head version because I need the wctdm driver for the 
 TDM04B (4*fxo modules)
 
 Does anybody know what is wrong?
 
  
 
 Thanks in advance!!
 
  
 
 Jairo Barahona Garita
 
 inCom Developer
---End of Original Message-


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Re: [Asterisk-Users] Extension Configuration Best Practice

[Rich Adamson]

 I would like to hear tips and tricks on extention config best practices, for
 example, naming, etc. and most of all, how to deal with extention that have
 a full time hardphone configured and assigned and then a softphone
 connecting to the same extention, for example, one employee has its
 hardphone on the office but sometimes when he travel, he uses his softphone
 to work with, what happens when two phones have the same user id and connect
 to the same asterisk? How are calls routed or how to handle this kind of
 scenarios.

In general terms and without being able to see how the extension is
defined in sip.conf, the last phone to register with * will get the
call.

Assuming both the hard and soft phones register every hour, it is
entirely possible the hard phone will get the call for the first
30 minutes and the soft phone for the next 30 minutes.


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[Asterisk-Users] PLEASE HELP X100P no responding

[Christian Callejon]

[EMAIL PROTECTED] ~]# modprobe zaptel[EMAIL PROTECTED] 
~]# modprobe wcfxoZT_CHANCONFIG failed on channel 1: No such device or 
address (6)FATAL: Error running install command for wcfxo



[EMAIL PROTECTED] ~]# ztcfg -vvv

Zaptel 
Configuration==Channel map:Channel 01: FXS 
Kewlstart (Default) (Slaves: 01)1 channels configured.ZT_CHANCONFIG 
failed on channel 1: No such device or address (6)

[EMAIL PROTECTED] /dev/zap]# ls -latotal 
0drwxr-xr-x 2 root 
root 120 jun 17 15:45 
.drwxr-xr-x 9 root 
root 5440 jun 17 15:45 
..crw-rw 1 asterisk asterisk 196, 254 jun 17 15:45 
channelcrw-rw 1 asterisk asterisk 196, 0 jun 17 15:45 
ctlcrw-rw 1 asterisk asterisk 196, 255 jun 17 15:45 
pseudocrw-rw 1 asterisk asterisk 196, 253 jun 17 15:45 
timer
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Re: [Asterisk-Users] Asterisk does not function without a DNS ser ver

[Rich Adamson]

Sure, put the entry in /etc/hosts. Check the /etc/host.conf to ensure
something like:
 order hosts,bind
is defined. That essentially tells your system to first check the
/etc/host for the host name and if not found use bind (dns).


 I'd really rather not run a DNS server if I don't have to. Surely ther
 must be a way to tell Asterisk not to rely on DNS?
 
 
 
 On 6/21/05, Guido Hecken [EMAIL PROTECTED] wrote:
   We have our Asterisk server running smoothly with a SIP BRI gateway
   for inbound calls. However if the Internet connection goes down and a
   DNS server becomes unreachable Asterisk basically does not function.
   By this I mean it does not answer call coming in from the gateway
   (which is on the local LAN) and you can't even reload it - just hangs
   there. If I change the DNS setting in resolv.conf to something else
   which is reachable all is well again.
  
   I have tried setting srvlookup=no in sip.conf but it made no difference.
  
   Does anyone know how I to make Asterisk continue working for local LAN
   users/gateways when a DNS server is not reachable?
  
  Try to use bind on the * Machine and configure it as a caching only
  nameserver.
  
  Hope, this helps
  
  Regards,
  
  Guido Hecken
 
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---End of Original Message-


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RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850

[John Cianfarani]

Does anyone know what the reason why Dell servers cause so many problems
for the digium hardware?
Better question any Dell models that don't have any these problems with
the digium hardware?

Thanks
John

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoa
Sent: Tuesday, June 21, 2005 3:40 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] ee1000 Ethernet in Dell 1850


Hmm, i dont think thats the reason they dont recommend the dell server.
The problems with the ee1000 kernel module are easily resolved, compile
the module into the kernel.

Zoa,

Andres wrote:




 Digium's site now lists the Dell 1850 as a potential problem server,
 as it uses the intel ee1000 Ethernet chipset (as do a majority of
 servers in the market!).

 To my knowledge, ALL dell servers with Gigabit interfaces now use
 the same chipset. Does this mean the Digium cards can't be used in
 Dell servers unless you disable the onboard ethernet?

 I don't want to disable the onboard interface, as I use the IPMI
 management facility for lights-out management. I have a 2850 that
 doesn't have any audio problems (the reason that I contacted Digium
 in the first place), so I'm wondering if Digium are simply guessing
 at problems.

 Does anyone know anything specific about the supposed
 incompatibilities with the ee1000 kernel module?


 I am not sure where you got that chipset reference but all our
 PowerEdge 1850s come with:
 Ethernet controller: Intel Corp. 82541GI/PI Gigabit Ethernet
Controller

 ...and they work fine with the TE410.


 There seems to be an ever-growing list of situations where you can't
 use the Digium cards. This is a concern to me.
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Re: [Asterisk-Users] How can you check that eg TDM04B hardware installed and drivers OK

[Rich Adamson]

 I am struggling to get my TDM04B working.  Just to rule out a hardware 
 problem how can I check 
that the hardware works?  How can I then
 check that the drivers are loaded correctly?
  

You didn't mention which linux distro you're using, so translate the
following into whatever your system expects. Try the following items:

1. from the linux command line, type 'dmesg' and look for
 Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules)
if you see that, the TDM card is recognized by the OS.

2. from the linux command line, type 'cat /proc/interrupts and look for
an entry with 'wctdm' in the list. If you don't see wctdm listed,
the module is not loaded as yet.

3. in /etc/zaptel.conf, ensure you have an entry like:
 fxsks=1-4 

4. if you're using a linux v2.6 kernel, read
 /usr/src/zaptel/README.udev

5. with asterisk stopped and from the linux command line, try
 sysconfig zaptel start

6. What do you see if you run 'zttool' from the linux command line?



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[Asterisk-Users] Call file calling twice

[Remco Barende]

Hi list!

The call files are working really great I just have one problem.

I am using this as my call file:
Channel: SIP/228
Context: from-internal
Extension: 0090
Priority: 1
Callerid:  0090

so the external number is connected to my sip phone. However after 
speaking for approx 30 seconds, Asterisk does a retry and I see the 
external number in my display on the second line. It does this on 
every call. When I'm finished I also see 2 records in the log files.


This is from the event log:
Jun 21 14:08:15 asterisk[1760]: Queued call to SIP/228 expired without 
completion after 0 attempt(s)

Jun 21 14:08:16 asterisk[1760]: Queued call to SIP/228 completed


Any idea why Asterisk is trying to place the call again even though the 
first attempt was succesfull and the call is still in progress?


I didn't specify a redial anywhere.

Thanks!


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[Asterisk-Users] Ground Start on Asterisk

[Syed Akbar]

Has anyone used ground start on Asterisk? I am using a TE110P connected a
Adtran 750 channel bank FXO card. It appears that Asterisk is not setting
the CAS RBS bits properly to seize the line. I can see this with the zttool
and the Adtran admin serial port status. I have configured the zaptel.conf
and Zapata.conf to use fxsgs on those channels.

Syed Akbar

Alico Systems Inc
www.alicosystems.com
Tel: 562-436-1510 


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RE: [Asterisk-Users] Looking for PRI Outbound Caller ID Configura tion

[Huddleston, Robert]

As an employee in the technical operations of a CLEC this information is easily 
obtainable by anyone that has access to the Class 5 switch servicing that 
PRI... A Q.931 trace in the Class 5 Switch will tell the whole story  

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Tuesday, June 21, 2005 8:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Looking for PRI Outbound Caller ID Configuration


 That didn't seem to work either.  Any other ideas?
 
 Thanks,
 
 Shaun
 
 -Original Message-
  I'm having trouble setting the outbound caller ID on calls I make 
  from my PRI trunk group.  The PRIs are served out of a 5ESS.  Telco 
  has set the
 PRIs
  up for user provided caller id information, so I believe I just 
  don't have it set up right in my dialplan or something.  I can't 
  seem to find an example of setting the outbound caller ID 
  specifically for a 5ESS.  Does anyone have an example configuration 
  that they have used with a 5ESS
 switch?
  Below is the my configuration from Zapata.conf and a sample 
  extension I've tried to use to connect a call with new caller ID 
  information provided by
 my
  PBX.  Any insight is most appreciated.
  
  [channels]
  priindication = outofband
  usecallerid=yes
  cidsignalling=bell
  hidecallerid=no
  callwaiting=no
  usecallingpres=yes
  callwaitingcallerid=no
  threewaycalling=yes
  transfer=yes
  cancallforward=yes
  callreturn=yes
  echocancel=yes
  echocancelwhenbridged=no
  rxgain=0.0
  txgain=0.0
  group=1
  callgroup=1
  pickupgroup=1
  immediate=no
  switchtype=5ess
  context=main
  signalling=pri_cpe
  group=1
  channel = 1-23
  channel = 25-47
  
  exten = 1234,1,Wait,1
  exten = 1234,2,Answer
  exten = 1234,3,SetCallerPres(allowed_passed_screen)
  exten = 1234,4,SetCIDNum(8881234567) exten = 
  1234,5,Dial(Zap/g1/18887654321,,,)
  exten = 1234,6,Hangup
  
 
 Try something like this...
 exten = _1NX,1,SetCallerID(8881234567|a)
 exten = _1NX,2,SetCIDName(MyName|a)
 exten = _1NX,3,Dial(ZAP/g1/${EXTEN})

If the above doesn't seem to work, the next step that I'd take is to have the 
central office tech's trace a call for you. They _can_ do that. 

The last pri that I implemented had similar problems, and it only took a few 
minutes with the right people on the phone to resolve it. Getting to the right 
person in the CO tends to be a problem in some cases. (In my case, they swore 
up and down things were configured correctly on their end. But after a couple 
of traces they found one of their switches was dropping callerid info.)


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[Asterisk-Users] Transfer

[Victor Alvarez]

Hi,
I'm afraid I don't know how to use 
thecommand Transfer. I have a couple of SIP users in the system and 
although exten = 35,1,Dial(SIP/33) works fine, exten = 35,1,Transfer(33) 
just don't work. All the description in the wiki is 'Transfer(exten)' without a 
single example.

35,1,Dial(SIP/33) would be a way to transfer 
the incoming call from 35 to 33, but what I want to do is to get 33 dialplan, 
not to dial 33. I mean, if 33 is 33,1,Voicemail that's what I would like to 
execute when calling 35.

Could anybody help me?

Thank you,
 Victor.
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RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 John Cianfarani
 Sent: Tuesday, June 21, 2005 8:16 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850
 
 Does anyone know what the reason why Dell servers cause so 
 many problems for the digium hardware?
 Better question any Dell models that don't have any these 
 problems with the digium hardware?

I've got a PowerEdge 1400SC (old, P4 1gHz, upgradable to dual proc)
that's been a absolute tank.  Got 2 TDM400P's in it and it supports a
very small office with mixed SIP and POTS inbound/outbound.  Running
Debian, of course.

Daryl
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RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850

[Dave Cotton]

On Tue, 2005-06-21 at 08:15 -0400, John Cianfarani wrote:
 Does anyone know what the reason why Dell servers cause so many problems
 for the digium hardware?
 Better question any Dell models that don't have any these problems with
 the digium hardware?

My experience it's not only Dell servers. I wanted to use a recent Dell
with 2 NICs i.e. the onboard and another, the second card was totally
hidden whichever PCI slot I used. Another Dell had a big red sticker
inside Do not update the BIOS.

In my days at NCR, people said we were Not Computers Really, I think
Dell is Discover Every Limitation Later.


-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Re: $0-per-month (pay as you go) provider with T.38?

[Joshua Colp]

I was just avoiding a potential nightmare when you tried to use a T.38
capable ATA to your provider through asterisk, and wondered why it didn't
work.

- Joshua Colp.


On 6/21/05 3:25 AM, Adam Megacz [EMAIL PROTECTED] wrote:

 
 Joshua Colp [EMAIL PROTECTED] writes:
 Make sure you're not using asterisk or you will have no T.38
 support, not even passthrough.
 
 Isn't this sort of like saying make sure you're not using Apache, or
 you will have no SMTP support?
 
 I call out to t38modem from an AGI script.  You know, that whole
 extensibility thing.
 
   - a
 
 
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RE: [Asterisk-Users] Transfer

[Nabeel Jafferali]

  35,1,Dial(SIP/33) would be a way to transfer the incoming call from 35 to
 33, but what I want to do is to get 33 dialplan, not to dial 33. I mean,
 if 33 is 33,1,Voicemail that's what I would like to execute when calling
 35.
 
 Could anybody help me?

Do you mean if you dial 35, you want Asterisk to run the 33 extensions
instead. If so, you need, for example:

exten = 35,1,Goto(33,1)

exten = 33,1,Voicemail

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990

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SV: SV: [Asterisk-Users] Presence and IM?

[Bjørn Ove Kristiansen]

Hello all!

First of all, thank you for all suggestions. As suggested, FOP does show
who's online, but it's not really what I'm looking for. As said before,
there's possibilities within the SIP protocol to have presence indication
(using SIMPLE?) and that's what I would like to use.

Not there yet, but imagine a small department with five staff members, all
equipped with laptops. Some of them are constantly on travel. With the
ability to use presence, any staff member will be able to tell right away
who's online and who's not, without going through an operator or opening up
FOP through their web browser. I'd consider this an advantage.

Regards,
Bjorn

-Opprinnelig melding-
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Juraj Bednar
Sendt: 19. juni 2005 19:19
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: SV: [Asterisk-Users] Presence and IM?

Hello,

  We have been running Asterisk for about a month now and one of the
  things I miss the most is the ability to se who's online and
  available and who's not. Surely, there's the manager interface, but
  unless you'd want to install extra software on each client computer,
  this is not a good option.
 
  Then there's the presence / IM function in SIP. Since we're only
  using SIP clients, this could easily solve some of our problems.
  However, I cannot get this to work with Asterisk using Eyebeam. Is
  this because the function is simply not supported within Asterisk?
 
  If lack of support is the case, anyone knows if this feature is to
  be implemented in the near future?

I have the same problem and am seeking for few weeks for a suitable
solution... If
you'll figure out something, please let me know.

 We use Polycom IP500s which when used with a 'hint' in extensions.conf,
 can show presence via the 'buddy list.'

could you post a snippet?

Does this hint work as a presence agent and sending notifies? Does
IM work with asterisk?

I would really like to support presence in Asterisk with Eyebeam as a
client. SIP Express
Router has this ability, but it's not a good choice either. Maybe it
would be possible to
port this feature from SER? 


  Juraj.
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Re: [Asterisk-Users] Transfer

[Jan Saell]

i know that there are extensive rework on the transfer in SIP at the moment.

--On Tuesday, June 21, 2005 13:40:47 +0100 Victor Alvarez 
[EMAIL PROTECTED] wrote:




Hi,
 I'm afraid I don't know how to use the command Transfer. I have a couple
of SIP users in the system and although exten = 35,1,Dial(SIP/33) works
fine, exten = 35,1,Transfer(33) just don't work. All the description in
the wiki is 'Transfer(exten)' without a single example.

 35,1,Dial(SIP/33) would be a way to transfer the incoming call from 35
to 33, but what I want to do is to get 33 dialplan, not to dial 33. I
mean, if 33 is 33,1,Voicemail that's what I would like to execute when
calling 35.

Could anybody help me?

Thank you,
  Victor.




--
+---
! Irial / YASK AB
! Att: Jan Saell
! Box 59, S-692 21 KUMLA, SWEDEN
! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05
! E-mail: [EMAIL PROTECTED]
! PGP Fingerprint: E957 23C8 9F51 0958 B9AD  7F18 404A 5DA1 F944 A08B


pgpmclaaGEcLY.pgp
Description: PGP signature
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[Asterisk-Users] ast_data help

[harry gaillac]

hello,

I need help with ast_data 
I downloaded asterisk from cvs
cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co
-r HEAD asterisk
and the latest ast_data.

When i run ./INSTALL.txt i get :

serveur1:/opt/asterisk/ast_data# ./INSTALL
patching file contrib/scripts/sip-friends.sql
patching file contrib/scripts/iax-friends.sql
patching file apps/app_voicemail.c
Hunk #1 succeeded at 27 with fuzz 1 (offset -18
lines).
Hunk #2 succeeded at 344 (offset 19 lines).
Hunk #3 succeeded at 614 (offset 19 lines).
Hunk #4 succeeded at 660 (offset 19 lines).
patching file apps/app_directory.c
Hunk #1 FAILED at 20.
Hunk #2 succeeded at 217 (offset 3 lines).
Hunk #3 succeeded at 319 (offset 3 lines).
Hunk #4 FAILED at 361.
2 out of 4 hunks FAILED -- saving rejects to file
apps/app_directory.c.rej
patching file channels/chan_sip.c
Hunk #1 succeeded at 30 with fuzz 2 (offset -29
lines).
Hunk #2 succeeded at 924 (offset 180 lines).
Hunk #3 succeeded at 1812 (offset 176 lines).
Hunk #4 succeeded at 1914 (offset 181 lines).
Hunk #5 succeeded at 1991 (offset 184 lines).
Hunk #6 succeeded at 2388 (offset 187 lines).
patching file channels/chan_iax2.c
Hunk #1 succeeded at 70 (offset 4 lines).
Hunk #2 succeeded at 735 (offset 18 lines).
Hunk #3 succeeded at 1086 (offset 18 lines).
Hunk #4 succeeded at 4975 (offset 21 lines).
Hunk #5 succeeded at 5790 (offset 21 lines).
patching file Makefile
Hunk #1 succeeded at 222 (offset -11 lines).
Hunk #2 succeeded at 253 with fuzz 2 (offset -10
lines).
patching file pbx.c
Hunk #1 FAILED at 21.
Hunk #2 succeeded at 22 with fuzz 2 (offset -19
lines).
Hunk #3 succeeded at 98 (offset 3 lines).
Hunk #4 succeeded at 776 (offset 27 lines).
Hunk #5 succeeded at 887 (offset 27 lines).
Hunk #6 succeeded at 901 (offset 27 lines).
Hunk #7 succeeded at 1720 (offset 27 lines).
Hunk #8 succeeded at 1732 (offset 27 lines).
Hunk #9 succeeded at 1769 (offset 27 lines).
Hunk #10 succeeded at 1809 (offset 27 lines).
Hunk #11 succeeded at 1838 (offset 27 lines).
Hunk #12 succeeded at 1868 (offset 27 lines).
1 out of 12 hunks FAILED -- saving rejects to file
pbx.c.rej
patching file asterisk.c
Hunk #1 FAILED at 57.
Hunk #2 succeeded at 2096 (offset 152 lines).
1 out of 2 hunks FAILED -- saving rejects to file
asterisk.c.rej

please to help me 

harry






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Re: [Asterisk-Users] Transfer

[sylvain garcia]

In your extension.conf

35,1Dial(SIP/33,Ttr)
in order to transfert during a call #33



Victor Alvarez a crit:

  
  
  
  Hi,
  I'm afraid I don't know how to use
thecommand Transfer. I have a couple of SIP users in the system and
although exten = 35,1,Dial(SIP/33) works fine, exten =
35,1,Transfer(33) just don't work. All the description in the wiki is
'Transfer(exten)' without a single example.
  
  35,1,Dial(SIP/33) would be a way to
transfer the incoming call from 35 to 33, but what I want to do is to
get 33 dialplan, not to dial 33. I mean, if 33 is 33,1,Voicemail that's
what I would like to execute when calling 35.
  
  Could anybody help me?
  
  Thank you,
   Victor.
  

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SV: SV: [Asterisk-Users] Presence and IM?

[Bjørn Ove Kristiansen]

Tried this, but unfortunately no luck.

Bjorn

-Opprinnelig melding-
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av
[EMAIL PROTECTED]
Sendt: 18. juni 2005 03:05
Til: asterisk-users@lists.digium.com
Emne: Re: SV: [Asterisk-Users] Presence and IM?

 
 We have been running Asterisk for about a month now and one of the 
 things I miss the most is the ability to se who’s online and 
 available and who’s not. Surely, there’s the manager interface, but 
 unless you’d want to install extra software on each client computer,
 this is not a good option.
 
 Then there’s the presence / IM function in SIP. Since we’re only 
 using SIP clients, this could easily solve some of our problems. 
 However, I cannot get this to work with Asterisk using Eyebeam. Is 
 this because the function is simply not supported within Asterisk?
 
 If lack of support is the case, anyone knows if this feature is to 
 be implemented in the near future?
 

We use Polycom IP500s which when used with a 'hint' in extensions.conf, 
can show presence via the 'buddy list.'

-Ron

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Re: [Asterisk-Users] Extension Configuration Best Practice

[Joshua Colp]

Rich is indeed correct, Asterisk does not yet support multiple registrations
for a single peer entry. Thus when you register the previous registration is
discarded and the new one is used. Thus like he said, the last one that
registered gets the call.

- Joshua Colp.


On 6/21/05 9:39 AM, Rich Adamson [EMAIL PROTECTED] wrote:

 I would like to hear tips and tricks on extention config best practices, for
 example, naming, etc. and most of all, how to deal with extention that have
 a full time hardphone configured and assigned and then a softphone
 connecting to the same extention, for example, one employee has its
 hardphone on the office but sometimes when he travel, he uses his softphone
 to work with, what happens when two phones have the same user id and connect
 to the same asterisk? How are calls routed or how to handle this kind of
 scenarios.
 
 In general terms and without being able to see how the extension is
 defined in sip.conf, the last phone to register with * will get the
 call.
 
 Assuming both the hard and soft phones register every hour, it is
 entirely possible the hard phone will get the call for the first
 30 minutes and the soft phone for the next 30 minutes.
 
 
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Re: [Asterisk-Users] Ground Start on Asterisk

[Doug Lytle]

Syed Akbar wrote:


Has anyone used ground start on Asterisk? I am using a TE110P connected a
Adtran 750 channel bank FXO card. It appears that Asterisk is not setting

 

I'm testing with a TE110P and a Adit 600 with GS.  So far, it appears to 
be working fine.


I'm using CVS HEAD CVS-v1-0-06/20-05-14:13:51

Doug

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Re: 回复： [Asterisk-Users] storing CDR records in a MySQL database

[Joseph]

I got it. 
The instructions on the wiki page is not that clear for somebody who
works with database once every three-years :-)

I've copied all the text from the frame to a text file and tried to
create the table but that didn't work.  In addition the user need
privilege Create the table in addition to Insert.
So copying only text that starts with: 
 CREATE TABLE cdr (
...
did the trick.
-- 
#Joseph


On Tue, 2005-06-21 at 16:20 +0800, Gary Li wrote:
 Hi,
 Using what database name is not the matter.
 You should configure the dbname and user password in
 the /etc/asterisk/cdr_mysql.conf.



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[Asterisk-Users] MFC R2 - Can this be solved???

[j_amorim]

Hi, 

MFC R2 - UniCall implementation. 

The * is configured to send a 1101 Idle signal: 

zaptel.conf 


span=3,1,0,cas,hdb3 
# 
cas=9-23:1101 
cas=25-39:1101 


But is sending 1001 Idle signal 

Can anyone send me a tip?? 

Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 1001  -  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12  - 1011  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 R2 prot. err. [1/4000/Idle  /Idle ] cause 
32773 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 1001  -  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:2865 handle_uc_event: 
Unicall/12 event Protocol failure 
-- Unicall/12 protocol error. Cause 32773 
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Re: [Asterisk-Users] ast_data help

[Joshua Colp]

The asterisk source code has changed so much that ast_data no longer patches
cleanly against it. You'll either need to get the person who made ast_data
to update it, or manually figure out what to patch and where. If you look at
the filenames mentioned (ie: app_directory.c.rej) you'll see what failed to
patch.

- Joshua Colp.


On 6/21/05 10:00 AM, harry gaillac [EMAIL PROTECTED] wrote:

 hello,
 
 I need help with ast_data
 I downloaded asterisk from cvs
 cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co
 -r HEAD asterisk
 and the latest ast_data.
 
 When i run ./INSTALL.txt i get :
 
 serveur1:/opt/asterisk/ast_data# ./INSTALL
 patching file contrib/scripts/sip-friends.sql
 patching file contrib/scripts/iax-friends.sql
 patching file apps/app_voicemail.c
 Hunk #1 succeeded at 27 with fuzz 1 (offset -18
 lines).
 Hunk #2 succeeded at 344 (offset 19 lines).
 Hunk #3 succeeded at 614 (offset 19 lines).
 Hunk #4 succeeded at 660 (offset 19 lines).
 patching file apps/app_directory.c
 Hunk #1 FAILED at 20.
 Hunk #2 succeeded at 217 (offset 3 lines).
 Hunk #3 succeeded at 319 (offset 3 lines).
 Hunk #4 FAILED at 361.
 2 out of 4 hunks FAILED -- saving rejects to file
 apps/app_directory.c.rej
 patching file channels/chan_sip.c
 Hunk #1 succeeded at 30 with fuzz 2 (offset -29
 lines).
 Hunk #2 succeeded at 924 (offset 180 lines).
 Hunk #3 succeeded at 1812 (offset 176 lines).
 Hunk #4 succeeded at 1914 (offset 181 lines).
 Hunk #5 succeeded at 1991 (offset 184 lines).
 Hunk #6 succeeded at 2388 (offset 187 lines).
 patching file channels/chan_iax2.c
 Hunk #1 succeeded at 70 (offset 4 lines).
 Hunk #2 succeeded at 735 (offset 18 lines).
 Hunk #3 succeeded at 1086 (offset 18 lines).
 Hunk #4 succeeded at 4975 (offset 21 lines).
 Hunk #5 succeeded at 5790 (offset 21 lines).
 patching file Makefile
 Hunk #1 succeeded at 222 (offset -11 lines).
 Hunk #2 succeeded at 253 with fuzz 2 (offset -10
 lines).
 patching file pbx.c
 Hunk #1 FAILED at 21.
 Hunk #2 succeeded at 22 with fuzz 2 (offset -19
 lines).
 Hunk #3 succeeded at 98 (offset 3 lines).
 Hunk #4 succeeded at 776 (offset 27 lines).
 Hunk #5 succeeded at 887 (offset 27 lines).
 Hunk #6 succeeded at 901 (offset 27 lines).
 Hunk #7 succeeded at 1720 (offset 27 lines).
 Hunk #8 succeeded at 1732 (offset 27 lines).
 Hunk #9 succeeded at 1769 (offset 27 lines).
 Hunk #10 succeeded at 1809 (offset 27 lines).
 Hunk #11 succeeded at 1838 (offset 27 lines).
 Hunk #12 succeeded at 1868 (offset 27 lines).
 1 out of 12 hunks FAILED -- saving rejects to file
 pbx.c.rej
 patching file asterisk.c
 Hunk #1 FAILED at 57.
 Hunk #2 succeeded at 2096 (offset 152 lines).
 1 out of 2 hunks FAILED -- saving rejects to file
 asterisk.c.rej
 
 please to help me
 
 harry
 
 
 
 
 
 
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Re: SV: SV: [Asterisk-Users] Presence and IM?

[Joshua Colp]

Asterisk does support the presence support in SIP, at least in CVS head. It
takes some fiddling to make it work. Below you'll find a link that will
hopefully help you. As for SIMPLE it's actually SIP's messaging protocol,
which Asterisk does not ... quite ... support.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extension
s - Details the hint priority, what it is - what it does and gives a link
to a scenario where a SNOM phone was used.

Please note that the source code mentioned on the See Also link is already
present in Asterisk. As well, the context where you put your hints needs to
be accessible to the SIP phone.

- Joshua Colp.


On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote:

 Hello all!
 
 First of all, thank you for all suggestions. As suggested, FOP does show
 who's online, but it's not really what I'm looking for. As said before,
 there's possibilities within the SIP protocol to have presence indication
 (using SIMPLE?) and that's what I would like to use.
 
 Not there yet, but imagine a small department with five staff members, all
 equipped with laptops. Some of them are constantly on travel. With the
 ability to use presence, any staff member will be able to tell right away
 who's online and who's not, without going through an operator or opening up
 FOP through their web browser. I'd consider this an advantage.
 
 Regards,
 Bjorn
 
 -Opprinnelig melding-
 Fra: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] På vegne av Juraj Bednar
 Sendt: 19. juni 2005 19:19
 Til: Asterisk Users Mailing List - Non-Commercial Discussion
 Emne: Re: SV: [Asterisk-Users] Presence and IM?
 
 Hello,
 
 We have been running Asterisk for about a month now and one of the
 things I miss the most is the ability to se who's online and
 available and who's not. Surely, there's the manager interface, but
 unless you'd want to install extra software on each client computer,
 this is not a good option.
 
 Then there's the presence / IM function in SIP. Since we're only
 using SIP clients, this could easily solve some of our problems.
 However, I cannot get this to work with Asterisk using Eyebeam. Is
 this because the function is simply not supported within Asterisk?
 
 If lack of support is the case, anyone knows if this feature is to
 be implemented in the near future?
 
 I have the same problem and am seeking for few weeks for a suitable
 solution... If
 you'll figure out something, please let me know.
 
 We use Polycom IP500s which when used with a 'hint' in extensions.conf,
 can show presence via the 'buddy list.'
 
 could you post a snippet?
 
 Does this hint work as a presence agent and sending notifies? Does
 IM work with asterisk?
 
 I would really like to support presence in Asterisk with Eyebeam as a
 client. SIP Express
 Router has this ability, but it's not a good choice either. Maybe it
 would be possible to
 port this feature from SER?
 
 
   Juraj.
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[Asterisk-Users] MFC R2 - Can this problem be solved??????????

[j_amorim]

Hi, 

MFC R2 - UniCall implementation. 

The * is configured to send a 1101 Idle signal: 

zaptel.conf 


span=3,1,0,cas,hdb3 
# 
cas=9-23:1101 
cas=25-39:1101 


But is sending 1001 Idle signal 

Can anyone send me a tip?? 

Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 1001  -  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12  - 1011  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 R2 prot. err. [1/4000/Idle  /Idle ] cause 
32773 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 1001  -  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:2865 handle_uc_event: 
Unicall/12 event Protocol failure 
-- Unicall/12 protocol error. Cause 32773 
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RE: [Asterisk-Users] Extension Configuration Best Practice

[Anton Krall]

In environments where users have their hard and soft phones... How do you
glue everything together? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Rich Adamson
|Sent: Martes, 21 de Junio de 2005 07:39 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
|
| I would like to hear tips and tricks on extention config best 
| practices, for example, naming, etc. and most of all, how to 
|deal with 
| extention that have a full time hardphone configured and 
|assigned and 
| then a softphone connecting to the same extention, for example, one 
| employee has its hardphone on the office but sometimes when 
|he travel, 
| he uses his softphone to work with, what happens when two 
|phones have 
| the same user id and connect to the same asterisk? How are calls 
| routed or how to handle this kind of scenarios.
|
|In general terms and without being able to see how the 
|extension is defined in sip.conf, the last phone to register 
|with * will get the call.
|
|Assuming both the hard and soft phones register every hour, it 
|is entirely possible the hard phone will get the call for the 
|first 30 minutes and the soft phone for the next 30 minutes.
|
|
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Re: [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.

[Rich Adamson]

 I have not really tried any other values for N1, M1 or CGM.  I actually
 used the formulas from the Si3035 data sheet to calculate what they
 should be for 8Khz.  There's a lot of math in there, but it looks like
 there may be several ways to arrive at the same output values.  Not sure
 if using a different calculation for the different dividers might give
 better results using the same crystal or not.  This was my first shot at
 it but your idea seems like a good one.

It might be possible to change the values slightly to judge their impact.
I've not done the math, so not sure if changing the values has any real
merit.

 I'm not sure what profiling tools might be useful, but would be delighted
 to hear any suggestions that anyone can contribute.  It really appears
 that things are choking up somewhere in the interrupt handling routines
 and I'm guessing somewhere in the zaptel driver.

I'm not a proficient programmer at all, but some experienced programmers
use various profiling tools to help understand which routines are consuming
cycles. It would seem like that could be used to help isolate the 
repetitive cpu spikes.
 
 If the problem turns out to be a timing sync problem due to oversampling a
 sample or so per second, then the best solution may be a hardware one. 

Its my understanding (which could be incorrect) the clock on the TDM card
is used for two purposes. First to drive the onboard chipset and second
to generate an interrupt on a recurring basis. And, that same interrupt is
used to time or sync other functions within asterisk. At least that
has been the argument behind do you have a zaptel timing device. Each
of the digium cards seem to use that same architecture, however it also
seems the TDM card is the only card that leaves something on the table.

So, is the missed data resulting from:
 1. pcm data arriving to fast/slow on the card for the pci controller to
cause an interrupt and transfer the data across the bus reliably?
 2. to much time spent handling the interrupt within asterisk drivers
causing an interrupt to be missed (or delayed service)?
 3. timing design conflicts between clocking the 3050 (pcm conversation)
verses interrupt requirements?
 4. potential problems in the pci controller design?

I would have to believe the clock is driving the pcm encoding function
within the 3050 chip, and the design objective is to cause the chip to
encode exactly 8,000 samples per second. Therefore, changing that 
clocking mechanism is likely to generate 7,990 or 8,010 samples (or
some other non-standard rate) that is likely to negatively impact other
asterisk functions (due to the reliance on the interrupts as a timing
source). But, the flip side of that would suggest the existing design
is running at some rate other then 8,000 samples/sec now.

For the TDM card, there is no such thing as syncing its clock to anything
since its handling incoming analog audio that contains no such info.

 I'm still trying to get a handle on exactly how the overall system timing
 works with the zaptel driver.  It does not seem like even multiple
 (non-t1) cards of the same type in an asterisk system sync their clocks. 
 For example, each seems to bring data into the system according to the
 timing of it's own internal oscillator.  

I believe that is correct and was very likely one of the driving forces
in the design of the TDM card (e.g., one interrupt handling four pstn
lines as opposed to multiple x100p cards each with their own interrupt
servicing requirements.

 That's my assessment of the wcfxo
 style cards at least.  The TDM400 seems to derive it's clock a little
 differently.  Perhaps somebody could jump in and shed a little light on
 how the hardware clocking works for that card.  It seems that overall the
 basic theory of operation is quite similar - Tiger Jet 320 PCI controller,
 DAA (or SLIC for FXS) etc.  As far as I know, the problems of CPU spikes
 and data loss are not apparent on a properly configured T1 setup.

I don't believe anyone has confirmed the cpu spikes are actually 
responsible for missed frames. At least I won't assume that for now.

The T1 card is different since a properly configured card will sync its
onboard clock with an external source that is considered highly accurate.
When the clock is in sync, there is no such thing as missed pcm frames
on a T1 card. But, I'm sure you're read the various postings from folks
that did not properly define the card sync and those postings generally
relate to audio clicks (and other disturbances) that are essentially the
same apparent issues as a free-wheeling TDM clock.

 I think that any data that we can gain from others running vmstat 1
 (looking for cpu spikes) in combination with running ztclock would be
 useful.  Especially on differing hardware including the various T1 cards. 
 ztclock is looking pretty good to me on my hardware, but as with most
 polling type tests I would anticipate there must be some margin of error. 
 I 

[Asterisk-Users] modprobe wctdm waiting for ever

[Edgardo Lust]

Hi,
I have a pentium 4 with Intel motherboard and one TDM400P (2 fxs, 2fxo)

modprobe zaptel  is Ok
but
When I execute modprobe wctdm never load the module, I can wait for 1
year but never response me (error or OK). I need to do ctrl+c 

Any idea?

Edgardo


 [EMAIL PROTECTED] 06/21/05 10:07 AM 
i know that there are extensive rework on the transfer in SIP at the
moment.

--On Tuesday, June 21, 2005 13:40:47 +0100 Victor Alvarez 
[EMAIL PROTECTED] wrote:


 Hi,
  I'm afraid I don't know how to use the command Transfer. I have a
couple
 of SIP users in the system and although exten = 35,1,Dial(SIP/33)
works
 fine, exten = 35,1,Transfer(33) just don't work. All the description
in
 the wiki is 'Transfer(exten)' without a single example.

  35,1,Dial(SIP/33) would be a way to transfer the incoming call from
35
 to 33, but what I want to do is to get 33 dialplan, not to dial 33. I
 mean, if 33 is 33,1,Voicemail that's what I would like to execute when
 calling 35.

 Could anybody help me?

 Thank you,
   Victor.



-- 
+---
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! Att: Jan Saell
! Box 59, S-692 21 KUMLA, SWEDEN
! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05
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RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850

[Rich Adamson]

 Does anyone know what the reason why Dell servers cause so many problems
 for the digium hardware?
 Better question any Dell models that don't have any these problems with
 the digium hardware?

I don't believe the problem is Dell servers as much as it is with
underlying linux/digium hardware limitations in terms of pci bus 
advancements, port  dma  interrupt mapping, etc, etc.

99% of the Dell hardware is sold in the Windows market space, and the
linux/digium space isn't exactly on track with the Windows hardware
changes that have occurred.


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Re: [Asterisk-Users] Ground Start on Asterisk

[Rich Adamson]

 Has anyone used ground start on Asterisk? I am using a TE110P connected a
 Adtran 750 channel bank FXO card. It appears that Asterisk is not setting
 
   
 
 I'm testing with a TE110P and a Adit 600 with GS.  So far, it appears to 
 be working fine.
 
 I'm using CVS HEAD CVS-v1-0-06/20-05-14:13:51

That's really odd. Never heard of CVS Head and CVS-v1-0-6 in the
same display. I'd guess you're running Stable v1.0.6.


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RE: [Asterisk-Users] Extension Configuration Best Practice

[Anton Krall]

Ok, so how are you guys coping with scenarios like this? Managers working in
the office during the day or mid day and then in the afternoon, working
remotely using their laptops? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Joshua Colp
|Sent: Martes, 21 de Junio de 2005 08:20 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
|
|Rich is indeed correct, Asterisk does not yet support multiple 
|registrations for a single peer entry. Thus when you register 
|the previous registration is discarded and the new one is 
|used. Thus like he said, the last one that registered gets the call.
|
|- Joshua Colp.
|
|
|On 6/21/05 9:39 AM, Rich Adamson [EMAIL PROTECTED] wrote:
|
| I would like to hear tips and tricks on extention config best 
| practices, for example, naming, etc. and most of all, how to deal 
| with extention that have a full time hardphone configured and 
| assigned and then a softphone connecting to the same extention, for 
| example, one employee has its hardphone on the office but sometimes 
| when he travel, he uses his softphone to work with, what 
|happens when 
| two phones have the same user id and connect to the same asterisk? 
| How are calls routed or how to handle this kind of scenarios.
| 
| In general terms and without being able to see how the extension is 
| defined in sip.conf, the last phone to register with * will get the 
| call.
| 
| Assuming both the hard and soft phones register every hour, it is 
| entirely possible the hard phone will get the call for the first 30 
| minutes and the soft phone for the next 30 minutes.
| 
| 
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Re: [Asterisk-Users] Extension Configuration Best Practice

[Joshua Colp]

Multiple entries in sip.conf, with a macro specifying multiple places to
call for extensions... That's what I do.

- Joshua Colp.


On 6/21/05 10:29 AM, Anton Krall [EMAIL PROTECTED] wrote:

 In environments where users have their hard and soft phones... How do you
 glue everything together?
 
 |-Original Message-
 |From: [EMAIL PROTECTED]
 |[mailto:[EMAIL PROTECTED] On Behalf Of
 |Rich Adamson
 |Sent: Martes, 21 de Junio de 2005 07:39 a.m.
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
 |
 | I would like to hear tips and tricks on extention config best
 | practices, for example, naming, etc. and most of all, how to
 |deal with 
 | extention that have a full time hardphone configured and
 |assigned and 
 | then a softphone connecting to the same extention, for example, one
 | employee has its hardphone on the office but sometimes when
 |he travel, 
 | he uses his softphone to work with, what happens when two
 |phones have 
 | the same user id and connect to the same asterisk? How are calls
 | routed or how to handle this kind of scenarios.
 |
 |In general terms and without being able to see how the
 |extension is defined in sip.conf, the last phone to register
 |with * will get the call.
 |
 |Assuming both the hard and soft phones register every hour, it
 |is entirely possible the hard phone will get the call for the
 |first 30 minutes and the soft phone for the next 30 minutes.
 |
 |
 |___
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 |Asterisk-Users@lists.digium.com
 |http://lists.digium.com/mailman/listinfo/asterisk-users
 |To UNSUBSCRIBE or update options visit:
 |   http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Cisco 7750

[Mark Johnson]

I have read of people attempting to do this, and I just wanted everyone 
to know about what we've discovered about the Cisco 7750.  If you don't 
know what it is, it's basically a blade server.  I have 1 power blade, 1 
alarm processor, 2 system processing engines and 1 multi-service route 
processor.  We just got asterisk running on this today!!!


We haven't tested the T1 with it, yet, but I pretty sure it will work 
OK.  All of the FX ports work beautifully right now.  The big deal about 
this for me is that I  have battled over and over again with interrupt 
issues with Digium hardware.  This is sweet because all the T1 
processing including echo cancellation should be done on the route 
processor.  Asterisk doesn't have to do much of anything.


Thought you guys might want to know.  I'll keep you posted as to how it 
works for us!!


Mark
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Re: [Asterisk-Users] How can you check that eg TDM04B hardware installed and drivers OK

[Mike M]

On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote:
 
  I am struggling to get my TDM04B working.  Just to rule out a hardware 
  problem how can I check 
 that the hardware works?  How can I then
  check that the drivers are loaded correctly?
   
 
 1. from the linux command line, type 'dmesg' and look for
  Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules)
 if you see that, the TDM card is recognized by the OS.
 

Here's what I get on a working system:

[EMAIL PROTECTED] src]# modprobe wctdm
[EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for device
01:0a.0
Jun 21 10:10:55 b2 kernel: Freshmaker version: 71
Jun 21 10:10:55 b2 kernel: Freshmaker passed register test
Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO
Jun 21 10:10:55 b2 kernel: Module 1: Not installed
Jun 21 10:10:55 b2 kernel: Module 2: Not installed
Jun 21 10:10:55 b2 kernel: Module 3: Not installed
Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV
E/F (4 modules)
Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States / North
America)

[EMAIL PROTECTED] src]# cat /proc/interrupts 
   CPU0   
  0:   17893766  XT-PIC  timer
  1:  2  XT-PIC  keyboard
  2:  0  XT-PIC  cascade
  4:  357411641  XT-PIC  eth0, wanpipe1
  8:  1  XT-PIC  rtc
 10:3381408  XT-PIC  Intel ICH2
 11:  178236906  XT-PIC  wctdm
 14:  50492  XT-PIC  ide0
 15:  0  XT-PIC  ide1
NMI:  0 
ERR:  0
[EMAIL PROTECTED] src]# cat /proc/interrupts 
   CPU0   
  0:   17894203  XT-PIC  timer
  1:  2  XT-PIC  keyboard
  2:  0  XT-PIC  cascade
  4:  357419974  XT-PIC  eth0, wanpipe1
  8:  1  XT-PIC  rtc
 10:3381408  XT-PIC  Intel ICH2
 11:  178241275  XT-PIC  wctdm
 14:  50494  XT-PIC  ide0
 15:  0  XT-PIC  ide1
NMI:  0 
ERR:  0

-- 
Mike
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[Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?

[Robert Rozman]

Hi,

I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, 
errors (duplicates) on more), when transferred inband from gsm gateway to NT 
port of quadbri under bristuffed Asterisk.


Since Asterisk is claimed to have good dtmf recognizer, I suspect there are 
some settings to workarouned... I've tried dtmf relax, but didn't help, so I 
suspect gain settings


Is there any other possible cause of unreliable dtmf inband recognition ? 
Where can I set gain on voice channel (I guess majority of settings under 
bristuff in zaptel.conf are dummy) ?


Any other advice on this problem or similar experience ?

Thanks in advance,

regards,

Rob.

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RE: [Asterisk-Users] Extension Configuration Best Practice

[Rich Adamson]

One method is to give each hard and soft phone their own extension
numbers. Then create a 'call forwarding' approach that essentially
says when someone dials x1234, ring 1235 instead. I believe there
are a couple of asterisk-based call forwarding approaches shown in
the wiki.

One such way is to have your user dial a predetermined extension
(eg, 4123) and the code within that extension definition does a
dbput of some value (eg, true, 1, or whatever). Then when someone
calls x1234, test the value using dbget to see if the call should
be forwarded. That's all hard-coded logic, but the wiki has some
macros to do that same kind of thing.


 In environments where users have their hard and soft phones... How do you
 glue everything together? 
 
 |-Original Message-
 |From: [EMAIL PROTECTED] 
 |[mailto:[EMAIL PROTECTED] On Behalf Of 
 |Rich Adamson
 |Sent: Martes, 21 de Junio de 2005 07:39 a.m.
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
 |
 | I would like to hear tips and tricks on extention config best 
 | practices, for example, naming, etc. and most of all, how to 
 |deal with 
 | extention that have a full time hardphone configured and 
 |assigned and 
 | then a softphone connecting to the same extention, for example, one 
 | employee has its hardphone on the office but sometimes when 
 |he travel, 
 | he uses his softphone to work with, what happens when two 
 |phones have 
 | the same user id and connect to the same asterisk? How are calls 
 | routed or how to handle this kind of scenarios.
 |
 |In general terms and without being able to see how the 
 |extension is defined in sip.conf, the last phone to register 
 |with * will get the call.
 |
 |Assuming both the hard and soft phones register every hour, it 
 |is entirely possible the hard phone will get the call for the 
 |first 30 minutes and the soft phone for the next 30 minutes.
 |
 |
 |___
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 |Asterisk-Users@lists.digium.com
 |http://lists.digium.com/mailman/listinfo/asterisk-users
 |To UNSUBSCRIBE or update options visit:
 |   http://lists.digium.com/mailman/listinfo/asterisk-users
 |
 
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---End of Original Message-


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SV: SV: SV: [Asterisk-Users] Presence and IM?

[Bjørn Ove Kristiansen]

Hello again!

As said below, this was already tried. However, it doesn't work.

I should add that I've gotten the hint function to work through the
management interface, so the syntax should be right. But for presence it's
not fully compatible with SIP devices and software such as EyeBeam.

Bjorn

-Opprinnelig melding-
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Joshua Colp
Sendt: 21. juni 2005 15:16
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: SV: SV: [Asterisk-Users] Presence and IM?

Asterisk does support the presence support in SIP, at least in CVS head. It
takes some fiddling to make it work. Below you'll find a link that will
hopefully help you. As for SIMPLE it's actually SIP's messaging protocol,
which Asterisk does not ... quite ... support.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extension
s - Details the hint priority, what it is - what it does and gives a link
to a scenario where a SNOM phone was used.

Please note that the source code mentioned on the See Also link is already
present in Asterisk. As well, the context where you put your hints needs to
be accessible to the SIP phone.

- Joshua Colp.


On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote:

 Hello all!
 
 First of all, thank you for all suggestions. As suggested, FOP does show
 who's online, but it's not really what I'm looking for. As said before,
 there's possibilities within the SIP protocol to have presence indication
 (using SIMPLE?) and that's what I would like to use.
 
 Not there yet, but imagine a small department with five staff members, all
 equipped with laptops. Some of them are constantly on travel. With the
 ability to use presence, any staff member will be able to tell right away
 who's online and who's not, without going through an operator or opening
up
 FOP through their web browser. I'd consider this an advantage.
 
 Regards,
 Bjorn
 
 -Opprinnelig melding-
 Fra: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] På vegne av Juraj Bednar
 Sendt: 19. juni 2005 19:19
 Til: Asterisk Users Mailing List - Non-Commercial Discussion
 Emne: Re: SV: [Asterisk-Users] Presence and IM?
 
 Hello,
 
 We have been running Asterisk for about a month now and one of the
 things I miss the most is the ability to se who's online and
 available and who's not. Surely, there's the manager interface, but
 unless you'd want to install extra software on each client computer,
 this is not a good option.
 
 Then there's the presence / IM function in SIP. Since we're only
 using SIP clients, this could easily solve some of our problems.
 However, I cannot get this to work with Asterisk using Eyebeam. Is
 this because the function is simply not supported within Asterisk?
 
 If lack of support is the case, anyone knows if this feature is to
 be implemented in the near future?
 
 I have the same problem and am seeking for few weeks for a suitable
 solution... If
 you'll figure out something, please let me know.
 
 We use Polycom IP500s which when used with a 'hint' in extensions.conf,
 can show presence via the 'buddy list.'
 
 could you post a snippet?
 
 Does this hint work as a presence agent and sending notifies? Does
 IM work with asterisk?
 
 I would really like to support presence in Asterisk with Eyebeam as a
 client. SIP Express
 Router has this ability, but it's not a good choice either. Maybe it
 would be possible to
 port this feature from SER?
 
 
   Juraj.
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RE: [Asterisk-Users] Extension Configuration Best Practice

[Goolsby, Daniel S (Daniel)]

Could you just configure the extention to be a ring group instead of an
actual extention, or ring queue.. then have his phone/laptop log in
whenever he's at the office/coffee shop?

I know AMP has the functionality, but I haven't gone behind the scenes
and looked at the sip.conf or extensions.conf to see what the script or
macro is doing in a ring group/queue.

Daniel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Tuesday, June 21, 2005 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Extension Configuration Best Practice

Ok, so how are you guys coping with scenarios like this? Managers
working in
the office during the day or mid day and then in the afternoon, working
remotely using their laptops? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Joshua Colp
|Sent: Martes, 21 de Junio de 2005 08:20 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
|
|Rich is indeed correct, Asterisk does not yet support multiple 
|registrations for a single peer entry. Thus when you register 
|the previous registration is discarded and the new one is 
|used. Thus like he said, the last one that registered gets the call.
|
|- Joshua Colp.
|
|
|On 6/21/05 9:39 AM, Rich Adamson [EMAIL PROTECTED] wrote:
|
| I would like to hear tips and tricks on extention config best 
| practices, for example, naming, etc. and most of all, how to deal 
| with extention that have a full time hardphone configured and 
| assigned and then a softphone connecting to the same extention, for 
| example, one employee has its hardphone on the office but sometimes 
| when he travel, he uses his softphone to work with, what 
|happens when 
| two phones have the same user id and connect to the same asterisk? 
| How are calls routed or how to handle this kind of scenarios.
| 
| In general terms and without being able to see how the extension is 
| defined in sip.conf, the last phone to register with * will get the 
| call.
| 
| Assuming both the hard and soft phones register every hour, it is 
| entirely possible the hard phone will get the call for the first 30 
| minutes and the soft phone for the next 30 minutes.
| 
| 
| ___
| Asterisk-Users mailing list
| Asterisk-Users@lists.digium.com
| http://lists.digium.com/mailman/listinfo/asterisk-users
| To UNSUBSCRIBE or update options visit:
|http://lists.digium.com/mailman/listinfo/asterisk-users
|
|
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SV: SV: [Asterisk-Users] Presence and IM?

[Bjørn Ove Kristiansen]

Hello all!

First of all, thank you for all suggestions. As suggested, FOP does show
who's online, but it's not really what I'm looking for. As said before,
there's possibilities within the SIP protocol to have presence indication
(using SIMPLE?) and that's what I would like to use.

Not there yet, but imagine a small department with five staff members, all
equipped with laptops. Some of them are constantly on travel. With the
ability to use presence, any staff member will be able to tell right away
who's online and who's not, without going through an operator or opening up
FOP through their web browser. I'd consider this an advantage.

Regards,
Bjorn

-Opprinnelig melding-
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Juraj Bednar
Sendt: 19. juni 2005 19:19
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: SV: [Asterisk-Users] Presence and IM?

Hello,

  We have been running Asterisk for about a month now and one of the
  things I miss the most is the ability to se who's online and
  available and who's not. Surely, there's the manager interface, but
  unless you'd want to install extra software on each client computer,
  this is not a good option.
 
  Then there's the presence / IM function in SIP. Since we're only
  using SIP clients, this could easily solve some of our problems.
  However, I cannot get this to work with Asterisk using Eyebeam. Is
  this because the function is simply not supported within Asterisk?
 
  If lack of support is the case, anyone knows if this feature is to
  be implemented in the near future?

I have the same problem and am seeking for few weeks for a suitable
solution... If
you'll figure out something, please let me know.

 We use Polycom IP500s which when used with a 'hint' in extensions.conf,
 can show presence via the 'buddy list.'

could you post a snippet?

Does this hint work as a presence agent and sending notifies? Does
IM work with asterisk?

I would really like to support presence in Asterisk with Eyebeam as a
client. SIP Express
Router has this ability, but it's not a good choice either. Maybe it
would be possible to
port this feature from SER? 


  Juraj.
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[Asterisk-Users] chan_unicall, bug in 1.0.X - 99% CPU

[Gerardo Perosio]

I have a problem with inbound and outbound calls in asterisk. I read previous 
thread, and then test new versions of unicall, but I don't have success, the 
problem persist. The system suffers CPU peeks with a single conversation, 
with an interval of 5 or 10 seconds. 
My system has:
asterisk 1.0.5
zaptel 1.0.4
libunicall 0.0.3-pre3
hardware: E100P

Also, I made traces with strace, but all looked ok.

unicall.conf:
===
[channels]
language=es
context=default
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancelwhenbridged=yes
callgroup=1
pickupgroup=1
protocolclass=mfcr2
protocolvariant=ar,20,4,3
protocolend=co
channel=1-15
channel=17-31
busydetect=yes

Thanks.
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Re: [Asterisk-Users] MFC R2 - Can this problem be solved??????????

[Steve Underwood]

j_amorim wrote:

Hi, 

MFC R2 - UniCall implementation. 

The * is configured to send a 1101 Idle signal: 

zaptel.conf 



span=3,1,0,cas,hdb3 
# 
cas=9-23:1101 
cas=25-39:1101



You seem to have configured 1101 as the blocking signal.



But is sending 1001 Idle signal 
 


1001 is the usual idle signal.

Can anyone send me a tip?? 

Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 1001  -  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12  - 1011  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 R2 prot. err. [1/4000/Idle  /Idle ] cause 
32773 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:704 unicall_report: MFC/R2 
UniCall/12 1001  -  [1/4000/Idle  /Idle ] 
Jun 20 19:06:56 WARNING[24118]: chan_unicall.c:2865 handle_uc_event: 
Unicall/12 event Protocol failure 
   -- Unicall/12 protocol error. Cause 32773 
 

The far ends seems to be acting like the China or Thailand protocols, 
but I guess from your email address you are in Brazil. A number of 
people use my R2 software Brazil, but this is the second time someone 
has reported this problem this week. Strange. I am currently changing 
the software to make it more flexible, and tolerate this kind of 
behaviour. It should be ready in a day or two.


Regards,
Steve

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Re: [Asterisk-Users] Automatic Agent Login

[Waldo Rubinstein]

I think that will only assign that member to the Queue, but the member will still need to log in.- WaldoOn Jun 20, 2005, at 7:42 PM, Dan Morin wrote: In the queues.conf file, under your queue you can add the following: member=sip/ExtensionNumber where ExtensionNumber is the extension.  Then they should always be part of the queue. Hope this helps.   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Hendrik Magilsen Sent: Monday, June 20, 2005 4:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Automatic Agent Login  Is there an easy way to automatically log agents in? We are using the queuing function to front end a main number without really using multiple agents.  The downside is during a restart, or system reboot someone must remember to log in the agent.  If I could incorporate it into a startup script it would be much more convenient.  I’ve done some looking around and see references to persistent logins but that seems to be on the development platform. Hendrik ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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RE: [Asterisk-Users] Extension Configuration Best Practice

[Rich Adamson]

In addition to the call forwarding approach noted in an earlier response,
you can also have both the hard and soft phones register as different
exetensions, then use something like:
 exten = 1234,2,Dial(SIP/1234SIP/1235,15) 
to ring both phones, and the first one to answer gets the call.


 Ok, so how are you guys coping with scenarios like this? Managers working in
 the office during the day or mid day and then in the afternoon, working
 remotely using their laptops? 
 
 |-Original Message-
 |From: [EMAIL PROTECTED] 
 |[mailto:[EMAIL PROTECTED] On Behalf Of 
 |Joshua Colp
 |Sent: Martes, 21 de Junio de 2005 08:20 a.m.
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
 |
 |Rich is indeed correct, Asterisk does not yet support multiple 
 |registrations for a single peer entry. Thus when you register 
 |the previous registration is discarded and the new one is 
 |used. Thus like he said, the last one that registered gets the call.
 |
 |- Joshua Colp.
 |
 |
 |On 6/21/05 9:39 AM, Rich Adamson [EMAIL PROTECTED] wrote:
 |
 | I would like to hear tips and tricks on extention config best 
 | practices, for example, naming, etc. and most of all, how to deal 
 | with extention that have a full time hardphone configured and 
 | assigned and then a softphone connecting to the same extention, for 
 | example, one employee has its hardphone on the office but sometimes 
 | when he travel, he uses his softphone to work with, what 
 |happens when 
 | two phones have the same user id and connect to the same asterisk? 
 | How are calls routed or how to handle this kind of scenarios.
 | 
 | In general terms and without being able to see how the extension is 
 | defined in sip.conf, the last phone to register with * will get the 
 | call.
 | 
 | Assuming both the hard and soft phones register every hour, it is 
 | entirely possible the hard phone will get the call for the first 30 
 | minutes and the soft phone for the next 30 minutes.
 | 
 | 
 | ___
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 | Asterisk-Users@lists.digium.com
 | http://lists.digium.com/mailman/listinfo/asterisk-users
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 |http://lists.digium.com/mailman/listinfo/asterisk-users
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---End of Original Message-


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Re: SV: SV: [Asterisk-Users] Presence and IM?

[Joshua Colp]

My client (Entourage) did a word wrap... Couldn't fit it all on one line.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extension
Try that ^^^

- Joshua Colp.


On 6/21/05 11:04 AM, Anton Krall [EMAIL PROTECTED] wrote:

  Page cannot be found
 
 |-Original Message-
 |From: [EMAIL PROTECTED]
 |[mailto:[EMAIL PROTECTED] On Behalf Of
 |Joshua Colp
 |Sent: Martes, 21 de Junio de 2005 08:16 a.m.
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: Re: SV: SV: [Asterisk-Users] Presence and IM?
 |
 |Asterisk does support the presence support in SIP, at least in
 |CVS head. It takes some fiddling to make it work. Below you'll
 |find a link that will hopefully help you. As for SIMPLE it's
 |actually SIP's messaging protocol, which Asterisk does not ...
 |quite ... support.
 |
 |http://www.voip-info.org/tiki-index.php?page=Asterisk%20standar
 d%20extension
 |s - Details the hint priority, what it is - what it does and
 |gives a link to a scenario where a SNOM phone was used.
 |
 |Please note that the source code mentioned on the See Also
 |link is already present in Asterisk. As well, the context
 |where you put your hints needs to be accessible to the SIP phone.
 |
 |- Joshua Colp.
 |
 |
 |On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote:
 |
 | Hello all!
 | 
 | First of all, thank you for all suggestions. As suggested, FOP does
 | show who's online, but it's not really what I'm looking for. As said
 | before, there's possibilities within the SIP protocol to
 |have presence 
 | indication (using SIMPLE?) and that's what I would like to use.
 | 
 | Not there yet, but imagine a small department with five
 |staff members, 
 | all equipped with laptops. Some of them are constantly on
 |travel. With 
 | the ability to use presence, any staff member will be able to tell
 | right away who's online and who's not, without going through an
 | operator or opening up FOP through their web browser. I'd
 |consider this an advantage.
 | 
 | Regards,
 | Bjorn
 | 
 | -Opprinnelig melding-
 | Fra: [EMAIL PROTECTED]
 | [mailto:[EMAIL PROTECTED] På vegne av Juraj
 | Bednar
 | Sendt: 19. juni 2005 19:19
 | Til: Asterisk Users Mailing List - Non-Commercial Discussion
 | Emne: Re: SV: [Asterisk-Users] Presence and IM?
 | 
 | Hello,
 | 
 | We have been running Asterisk for about a month now and one of the
 | things I miss the most is the ability to se who's online and
 | available and who's not. Surely, there's the manager
 |interface, but 
 | unless you'd want to install extra software on each client
 |computer, 
 | this is not a good option.
 | 
 | Then there's the presence / IM function in SIP. Since we're only
 | using SIP clients, this could easily solve some of our problems.
 | However, I cannot get this to work with Asterisk using Eyebeam. Is
 | this because the function is simply not supported within Asterisk?
 | 
 | If lack of support is the case, anyone knows if this feature is to
 | be implemented in the near future?
 | 
 | I have the same problem and am seeking for few weeks for a suitable
 | solution... If you'll figure out something, please let me know.
 | 
 | We use Polycom IP500s which when used with a 'hint' in
 | extensions.conf, can show presence via the 'buddy list.'
 | 
 | could you post a snippet?
 | 
 | Does this hint work as a presence agent and sending notifies? Does
 | IM work with asterisk?
 | 
 | I would really like to support presence in Asterisk with
 |Eyebeam as a 
 | client. SIP Express Router has this ability, but it's not a good
 | choice either. Maybe it would be possible to port this feature from
 | SER?
 | 
 | 
 |   Juraj.
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RE: [Asterisk-Users] app_changrab.c released on pbxfreeware.org

[Anton Krall]

Where can We get it from? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Brian West
|Sent: Martes, 21 de Junio de 2005 09:11 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: [Asterisk-Users] app_changrab.c released on pbxfreeware.org
|
|I released app_changrab.c lastnight really late... It includes 
|a way to hijack a channel and originate calls from the CLI.
|
|/b
|---
|Keep Your Friends Close, But Your Enemies Even Closer...
|
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Re: [Asterisk-Users] ast_data help

[Matthew Boehm]

Or, you could just use the built-in database methods called RealTime.

Check out the wiki.

-Matthew

Joshua Colp wrote:

The asterisk source code has changed so much that ast_data no longer patches
cleanly against it. You'll either need to get the person who made ast_data
to update it, or manually figure out what to patch and where. If you look at
the filenames mentioned (ie: app_directory.c.rej) you'll see what failed to
patch.

- Joshua Colp.



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Re: [Asterisk-Users] Cisco 7750

[Trey Scarborough]

- Original Message - 
From: Mark Johnson [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 8:56 AM
Subject: [Asterisk-Users] Cisco 7750


I have read of people attempting to do this, and I just wanted everyone to 
know about what we've discovered about the Cisco 7750.  If you don't know 
what it is, it's basically a blade server.  I have 1 power blade, 1 alarm 
processor, 2 system processing engines and 1 multi-service route processor. 
We just got asterisk running on this today!!!


Just dont let cisco know

We haven't tested the T1 with it, yet, but I pretty sure it will work OK. 
All of the FX ports work beautifully right now.  The big deal about this 
for me is that I  have battled over and over again with interrupt issues 
with Digium hardware.  This is sweet because all the T1 processing 
including echo cancellation should be done on the route processor. 
Asterisk doesn't have to do much of anything.




so im guessing that all of the t1/fx ports are configured in the system 
processor and just talk sip/mgcp to the route proccessor.


That sounds like a pretty sweet setup If you could only get cisco to sell 
you the hardware without having to buy the software. 



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Re: [Asterisk-Users] Ground Start on Asterisk

[Doug Lytle]

Rich Adamson wrote:


Has anyone used ground start on Asterisk? I am using a TE110P connected a
Adtran 750 channel bank FXO card. It appears that Asterisk is not setting



 

I'm testing with a TE110P and a Adit 600 with GS.  So far, it appears to 
be working fine.


I'm using CVS HEAD CVS-v1-0-06/20-05-14:13:51
   



That's really odd. Never heard of CVS Head and CVS-v1-0-6 in the
same display. I'd guess you're running Stable v1.0.6.


 


I was wondering that myself.  I ran:

cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds


Did a make clean;make;make install
On all directories and started it back up. Fixed my queue MOH problem 
though, so figured I was reading it wrong



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Re: [Asterisk-Users] modprobe wctdm waiting for ever

[Mike M]

On Tue, Jun 21, 2005 at 10:47:59AM -0300, Edgardo Lust wrote:
 Hi,
 I have a pentium 4 with Intel motherboard and one TDM400P (2 fxs, 2fxo)
 
 modprobe zaptel  is Ok
 but
 When I execute modprobe wctdm never load the module, I can wait for 1
 year but never response me (error or OK). I need to do ctrl+c 

This sounds like a problem I had.  Some documentation out there is not
up to date with the newer line interface cards.

Try starting over:
modprobe -r wctdm
modprobe -r zaptel

Then just do:
modprobe wctdm

-- 
Mike
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[Asterisk-Users] Asterisk died - exactly every 60 minutes

[Ronald Wiplinger]

I have now a very strange situation.

Asterisk diesexactly every hour at hour:09  !!!

crontab has the entry:
10 3 * * * root /usr/sbin/asterisk-restart /dev/null 21

It seems that there is a time difference between mail server and 
asterisk server so that it might be synchronized to the crontab entry.


*CLI show version
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 
2005-06-18 14:53:44


I don't know anymore where to look at, and how to track this down.

Can anybody give me a hint???


bye

Ronald

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Re: [Asterisk-Users] ast_data help

[harry gaillac]

Hello,

Thanks for help i'll do it however i need same
examples 
to configure ast_data .

Harry
--- Joshua Colp [EMAIL PROTECTED] a écrit :

 The asterisk source code has changed so much that
 ast_data no longer patches
 cleanly against it. You'll either need to get the
 person who made ast_data
 to update it, or manually figure out what to patch
 and where. If you look at
 the filenames mentioned (ie: app_directory.c.rej)
 you'll see what failed to
 patch.
 
 - Joshua Colp.
 
 
 On 6/21/05 10:00 AM, harry gaillac
 [EMAIL PROTECTED] wrote:
 
  hello,
  
  I need help with ast_data
  I downloaded asterisk from cvs
  cvs -d
 :pserver:[EMAIL PROTECTED]:/usr/cvsroot co
  -r HEAD asterisk
  and the latest ast_data.
  
  When i run ./INSTALL.txt i get :
  
  serveur1:/opt/asterisk/ast_data# ./INSTALL
  patching file contrib/scripts/sip-friends.sql
  patching file contrib/scripts/iax-friends.sql
  patching file apps/app_voicemail.c
  Hunk #1 succeeded at 27 with fuzz 1 (offset -18
  lines).
  Hunk #2 succeeded at 344 (offset 19 lines).
  Hunk #3 succeeded at 614 (offset 19 lines).
  Hunk #4 succeeded at 660 (offset 19 lines).
  patching file apps/app_directory.c
  Hunk #1 FAILED at 20.
  Hunk #2 succeeded at 217 (offset 3 lines).
  Hunk #3 succeeded at 319 (offset 3 lines).
  Hunk #4 FAILED at 361.
  2 out of 4 hunks FAILED -- saving rejects to file
  apps/app_directory.c.rej
  patching file channels/chan_sip.c
  Hunk #1 succeeded at 30 with fuzz 2 (offset -29
  lines).
  Hunk #2 succeeded at 924 (offset 180 lines).
  Hunk #3 succeeded at 1812 (offset 176 lines).
  Hunk #4 succeeded at 1914 (offset 181 lines).
  Hunk #5 succeeded at 1991 (offset 184 lines).
  Hunk #6 succeeded at 2388 (offset 187 lines).
  patching file channels/chan_iax2.c
  Hunk #1 succeeded at 70 (offset 4 lines).
  Hunk #2 succeeded at 735 (offset 18 lines).
  Hunk #3 succeeded at 1086 (offset 18 lines).
  Hunk #4 succeeded at 4975 (offset 21 lines).
  Hunk #5 succeeded at 5790 (offset 21 lines).
  patching file Makefile
  Hunk #1 succeeded at 222 (offset -11 lines).
  Hunk #2 succeeded at 253 with fuzz 2 (offset -10
  lines).
  patching file pbx.c
  Hunk #1 FAILED at 21.
  Hunk #2 succeeded at 22 with fuzz 2 (offset -19
  lines).
  Hunk #3 succeeded at 98 (offset 3 lines).
  Hunk #4 succeeded at 776 (offset 27 lines).
  Hunk #5 succeeded at 887 (offset 27 lines).
  Hunk #6 succeeded at 901 (offset 27 lines).
  Hunk #7 succeeded at 1720 (offset 27 lines).
  Hunk #8 succeeded at 1732 (offset 27 lines).
  Hunk #9 succeeded at 1769 (offset 27 lines).
  Hunk #10 succeeded at 1809 (offset 27 lines).
  Hunk #11 succeeded at 1838 (offset 27 lines).
  Hunk #12 succeeded at 1868 (offset 27 lines).
  1 out of 12 hunks FAILED -- saving rejects to file
  pbx.c.rej
  patching file asterisk.c
  Hunk #1 FAILED at 57.
  Hunk #2 succeeded at 2096 (offset 152 lines).
  1 out of 2 hunks FAILED -- saving rejects to file
  asterisk.c.rej
  
  please to help me
  
  harry
  
  
  
  
  
  
 

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RE: [Asterisk-Users] Extension Configuration Best Practice

[Anton Krall]

Thx for your comments Guys, seems that is the logical way to go for now. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Joshua Colp
|Sent: Martes, 21 de Junio de 2005 09:18 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
|
|Multiple entries in sip.conf, with a macro specifying multiple 
|places to call for extensions... That's what I do.
|
|- Joshua Colp.
|
|
|On 6/21/05 10:29 AM, Anton Krall 
|[EMAIL PROTECTED] wrote:
|
| In environments where users have their hard and soft 
|phones... How do 
| you glue everything together?
| 
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of Rich 
| |Adamson
| |Sent: Martes, 21 de Junio de 2005 07:39 a.m.
| |To: Asterisk Users Mailing List - Non-Commercial Discussion
| |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
| |
| | I would like to hear tips and tricks on extention config best 
| | practices, for example, naming, etc. and most of all, how to
| |deal with
| | extention that have a full time hardphone configured and
| |assigned and
| | then a softphone connecting to the same extention, for 
|example, one 
| | employee has its hardphone on the office but sometimes when
| |he travel,
| | he uses his softphone to work with, what happens when two
| |phones have
| | the same user id and connect to the same asterisk? How are calls 
| | routed or how to handle this kind of scenarios.
| |
| |In general terms and without being able to see how the extension is 
| |defined in sip.conf, the last phone to register with * will get the 
| |call.
| |
| |Assuming both the hard and soft phones register every hour, it is 
| |entirely possible the hard phone will get the call for the first 30 
| |minutes and the soft phone for the next 30 minutes.
| |
| |
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[Asterisk-Users] Re: SNOM, Asterisk and Attended transfer (bug?)

[Steve Davies]

On 6/13/05, Steve Davies [EMAIL PROTECTED] wrote:
 Hi,
 
 I am using a number of snom190 phones, and an asterisk gateway
 server, and recently started experimenting with call transfers. The
 snom phones provide support for attended and un-attended call
 transfer, so I would rather use that than call-parking.
 
 I have found that un-attended transfer works fine, and that attended
 transfer works fine if the originating phone call is NON-SIP (ie.
 ISDN)
 
[snip]

The attended transfer problem is solved :) It turns out that an
attended transfer results in a different path for the RTP packets, and
it was our firewall rules not-expecting this behaviour that defeated
everything. *blush*

Thanks for all of the help and suggestions in the meantime.
Regards,
Steve
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Re: [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?

[Steve Underwood]

Robert Rozman wrote:


Hi,

I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, 
errors (duplicates) on more), when transferred inband from gsm gateway 
to NT port of quadbri under bristuffed Asterisk.


Since Asterisk is claimed to have good dtmf recognizer, I suspect 
there are some settings to workarouned... I've tried dtmf relax, but 
didn't help, so I suspect gain settings


Is there any other possible cause of unreliable dtmf inband 
recognition ? Where can I set gain on voice channel (I guess majority 
of settings under bristuff in zaptel.conf are dummy) ?


Any other advice on this problem or similar experience ?

Thanks in advance,


I kind of amazed if works at all when getting DTMF out of a GSM phone. 
You really shouldn't expect it to.


Regards,
Steve

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RE: [Asterisk-Users] Extension Configuration Best Practice

[Anton Krall]

I guess I would need to do something like that and mix with dialing 2
extension at the same time with dial(ext1exte2)

Seems the easier way to do it for now. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Rich Adamson
|Sent: Martes, 21 de Junio de 2005 10:25 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Extension Configuration Best Practice
|
|One method is to give each hard and soft phone their own 
|extension numbers. Then create a 'call forwarding' approach 
|that essentially says when someone dials x1234, ring 1235 
|instead. I believe there are a couple of asterisk-based call 
|forwarding approaches shown in the wiki.
|
|One such way is to have your user dial a predetermined 
|extension (eg, 4123) and the code within that extension 
|definition does a dbput of some value (eg, true, 1, or 
|whatever). Then when someone calls x1234, test the value using 
|dbget to see if the call should be forwarded. That's all 
|hard-coded logic, but the wiki has some macros to do that same 
|kind of thing.
|
|
| In environments where users have their hard and soft 
|phones... How do 
| you glue everything together?
| 
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of Rich 
| |Adamson
| |Sent: Martes, 21 de Junio de 2005 07:39 a.m.
| |To: Asterisk Users Mailing List - Non-Commercial Discussion
| |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice
| |
| | I would like to hear tips and tricks on extention config best 
| | practices, for example, naming, etc. and most of all, how to
| |deal with
| | extention that have a full time hardphone configured and
| |assigned and
| | then a softphone connecting to the same extention, for 
|example, one 
| | employee has its hardphone on the office but sometimes when
| |he travel,
| | he uses his softphone to work with, what happens when two
| |phones have
| | the same user id and connect to the same asterisk? How are calls 
| | routed or how to handle this kind of scenarios.
| |
| |In general terms and without being able to see how the extension is 
| |defined in sip.conf, the last phone to register with * will get the 
| |call.
| |
| |Assuming both the hard and soft phones register every hour, it is 
| |entirely possible the hard phone will get the call for the first 30 
| |minutes and the soft phone for the next 30 minutes.
| |
| |
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|
|---End of Original Message-
|
|
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RE: [Asterisk-Users] app_changrab.c released on pbxfreeware.org

[Dave Cotton]

On Tue, 2005-06-21 at 10:20 -0500, Anton Krall wrote:
 Where can We get it from? 
 
 |-Original Message-
 |From: [EMAIL PROTECTED] 
 |[mailto:[EMAIL PROTECTED] On Behalf Of 
 |Brian West
 |Sent: Martes, 21 de Junio de 2005 09:11 a.m.
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: [Asterisk-Users] app_changrab.c released on pbxfreeware.org
 |
 |I released app_changrab.c lastnight really late... It includes 
 |a way to hijack a channel and originate calls from the CLI.

Perhaps the subject line above tells you.


-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Extension Configuration Best Practice

[steve szmidt]

On Tuesday 21 June 2005 10:38, Goolsby, Daniel S (Daniel) wrote:
 Could you just configure the extention to be a ring group instead of an
 actual extention, or ring queue.. then have his phone/laptop log in
 whenever he's at the office/coffee shop?

As someone else pointed out if you want to keep it simple just use:

exten = 1234,2,Dial(SIP/1234SIP/1235,15) 

It will dial all their extensions. Why make it more complex?


-- 

Steve Szmidt

They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety.
Benjamin Franklin
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[Asterisk-Users] MeetMe Problems

[Waldo Rubinstein]

I have two asterisk machines. One of them has a Digium board (server  
A) and the other is simply using ztdummy (server B). Server A is  
running on Debian and Server B is running Gentoo. Server A is running  
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running  
Asterisk 1.0.7.


The problem I have is that when I try to transfer a call into a  
meetme room in server B, it simply hangs up the call. To be specific,  
when I press transfer (XFER on the Uniden UIP200) and then the meetme  
room number, the meetme room answers (I hear MOH), but when I hang  
up, it drops all calls and not just transfers the call to the meetme  
room.


Now, if I configure the meetme rooms indentically in server A, I can  
transfer the calls from server B to server A's meetme room and  
everything works just fine.


I would like for the meetme rooms to work in server B and not having  
to depend on server A for it.


Can anyone shed some light into why this is happening and, more  
importantly, how to fix it?


Thanks,
Waldo
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[Asterisk-Users] Best Echo Canceller.

[Chris Modesitt]

I know this is slight OT however I have decided that I need to but in some
echo cancellers on my PRI's.  I was wondering if anybody else was using a
hardware echo canceller capable of 24 T1's, how well it works and an
approximate price range:)

Thanks

Chris

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Re: [Asterisk-Users] Asterisk died - exactly every 60 minutes

[Sebastian Silva]

To see the entire log, in logger.conf:

full = notice,warning,error,debug,verbose

then:

tail -f /var/log/asterisk/full

in other console run asterisk, you will see all log output in the 
previous console and why asterisk stops.


Sebas

Ronald Wiplinger wrote:

I have now a very strange situation.

Asterisk diesexactly every hour at hour:09  !!!

crontab has the entry:
10 3 * * * root /usr/sbin/asterisk-restart /dev/null 21

It seems that there is a time difference between mail server and 
asterisk server so that it might be synchronized to the crontab entry.


*CLI show version
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 
2005-06-18 14:53:44


I don't know anymore where to look at, and how to track this down.

Can anybody give me a hint???


bye

Ronald

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--
Sebastian Silva
G R U P O  G A U S S
Depto. Sistemas
Av. Libertador 6250 4 piso
Tl.: 4 706- (int. 121)
[EMAIL PROTECTED]
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[Asterisk-Users] asterisk-api

[gale81]

Hi
I try to create a sip client with asterisk-api package,
I've a question:
 I can create a channel sip that generate sip signaling with Class Channel
or with another
class ?
Thanks Ale

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RE: SV: SV: [Asterisk-Users] Presence and IM?

[Anton Krall]

Can this hint system be used for gxp2000 phones or just for snoms? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Joshua Colp
|Sent: Martes, 21 de Junio de 2005 10:03 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: SV: SV: [Asterisk-Users] Presence and IM?
|
|My client (Entourage) did a word wrap... Couldn't fit it all 
|on one line.
|http://www.voip-info.org/tiki-index.php?page=Asterisk%20standar
d%20extension
|Try that ^^^
|
|- Joshua Colp.
|
|
|On 6/21/05 11:04 AM, Anton Krall 
|[EMAIL PROTECTED] wrote:
|
|  Page cannot be found
| 
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of
| |Joshua Colp
| |Sent: Martes, 21 de Junio de 2005 08:16 a.m.
| |To: Asterisk Users Mailing List - Non-Commercial Discussion
| |Subject: Re: SV: SV: [Asterisk-Users] Presence and IM?
| |
| |Asterisk does support the presence support in SIP, at least in
| |CVS head. It takes some fiddling to make it work. Below you'll
| |find a link that will hopefully help you. As for SIMPLE it's
| |actually SIP's messaging protocol, which Asterisk does not ...
| |quite ... support.
| |
| |http://www.voip-info.org/tiki-index.php?page=Asterisk%20standar
| d%20extension
| |s - Details the hint priority, what it is - what it does and
| |gives a link to a scenario where a SNOM phone was used.
| |
| |Please note that the source code mentioned on the See Also
| |link is already present in Asterisk. As well, the context
| |where you put your hints needs to be accessible to the SIP phone.
| |
| |- Joshua Colp.
| |
| |
| |On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote:
| |
| | Hello all!
| | 
| | First of all, thank you for all suggestions. As 
|suggested, FOP does
| | show who's online, but it's not really what I'm looking 
|for. As said
| | before, there's possibilities within the SIP protocol to
| |have presence 
| | indication (using SIMPLE?) and that's what I would like to use.
| | 
| | Not there yet, but imagine a small department with five
| |staff members, 
| | all equipped with laptops. Some of them are constantly on
| |travel. With 
| | the ability to use presence, any staff member will be able to tell
| | right away who's online and who's not, without going through an
| | operator or opening up FOP through their web browser. I'd
| |consider this an advantage.
| | 
| | Regards,
| | Bjorn
| | 
| | -Opprinnelig melding-
| | Fra: [EMAIL PROTECTED]
| | [mailto:[EMAIL PROTECTED] På vegne av Juraj
| | Bednar
| | Sendt: 19. juni 2005 19:19
| | Til: Asterisk Users Mailing List - Non-Commercial Discussion
| | Emne: Re: SV: [Asterisk-Users] Presence and IM?
| | 
| | Hello,
| | 
| | We have been running Asterisk for about a month now and 
|one of the
| | things I miss the most is the ability to se who's online and
| | available and who's not. Surely, there's the manager
| |interface, but 
| | unless you'd want to install extra software on each client
| |computer, 
| | this is not a good option.
| | 
| | Then there's the presence / IM function in SIP. Since we're only
| | using SIP clients, this could easily solve some of our problems.
| | However, I cannot get this to work with Asterisk using 
|Eyebeam. Is
| | this because the function is simply not supported 
|within Asterisk?
| | 
| | If lack of support is the case, anyone knows if this 
|feature is to
| | be implemented in the near future?
| | 
| | I have the same problem and am seeking for few weeks for 
|a suitable
| | solution... If you'll figure out something, please let me know.
| | 
| | We use Polycom IP500s which when used with a 'hint' in
| | extensions.conf, can show presence via the 'buddy list.'
| | 
| | could you post a snippet?
| | 
| | Does this hint work as a presence agent and sending 
|notifies? Does
| | IM work with asterisk?
| | 
| | I would really like to support presence in Asterisk with
| |Eyebeam as a 
| | client. SIP Express Router has this ability, but it's not a good
| | choice either. Maybe it would be possible to port this 
|feature from
| | SER?
| | 
| | 
| |   Juraj.
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| | 
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| |
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[Asterisk-Users] communication between IAX softphones

[Marco Parmeggiani]

I tried with several iax softphones:
iaxcomm
idefix
iaxphone

and i have a problems that i do not have with SIP clients.

A calls B, B phone starts ringing, asterisk says that call has been 
accepted, that is ringing but it is not yet answered. If B picks up, 
asterisk says that call has been answered but, *before* User B pick up, 
he is already able to hear User A and viceversa.


ciao
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RE: [Asterisk-Users] app_changrab.c released on pbxfreeware.org

[Anton Krall]

Outlook cut the subject... Damn MS.. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Dave Cotton
|Sent: Martes, 21 de Junio de 2005 11:28 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] app_changrab.c released on 
|pbxfreeware.org
|
|On Tue, 2005-06-21 at 10:20 -0500, Anton Krall wrote:
| Where can We get it from? 
| 
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of Brian 
| |West
| |Sent: Martes, 21 de Junio de 2005 09:11 a.m.
| |To: Asterisk Users Mailing List - Non-Commercial Discussion
| |Subject: [Asterisk-Users] app_changrab.c released on pbxfreeware.org
| |
| |I released app_changrab.c lastnight really late... It 
|includes a way 
| |to hijack a channel and originate calls from the CLI.
|
|Perhaps the subject line above tells you.
|
|
|--
|Dave Cotton [EMAIL PROTECTED]
|
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Re: [Asterisk-Users] accountcode not present in cdr

[Max Clark]

But what do people do with large LCR rules... Build special contexts for 
each peer/user and then include the main LCR context? This seems a 
little cludgy.


Is there any way to have the dialplan context set the account for cdr 
based on the accountcode defined in the sip.conf? At least this way I 
could have a single, generic dialplan.


-Max

Andres wrote:



Max Clark wrote:


Hi all,

I have what I hope will be a simple problem. In my sip.conf I have 
defined the accountcode field (see below), and they system does not 
report any errors when I reload the configuration. However when I look 
at my cdr detail (either the csv on disk, or the mysql info) the 
accountcode that I have specified is missing. I have scoured the list 
and have seen a few postings on this with no solutions. What should I 
be looking at to debug this.


Thanks in advance,
Max



Setting the accountcode in sip.conf is totally unreliable.  It does not 
work in many cases.  Your best bet is to set it in a context via the 
command:

SetAccount([account]):  Set  the  channel account code for billing



I am running Asterisk 1.0.7 on CentOS 4.0:
Linux 2.6.9-11.EL #1 Wed Jun 8 16:59:52 CDT 2005 i686 i686 i386 GNU/Linux

[provider1]
accountcode=provider1
type=friend
host=10.1.1.1
dtmfmode=rfc2833
username=user
secret=12345
qualify=no
canreinvite=no
insecure=very
disallow=all
allow=ulaw
allow=gsm

[provider2]
type=peer
accountcode=provider2
secret=54321
username=user
host=10.1.1.10
dtmfmode=rfc2833






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Re: [Asterisk-Users] Cisco 7750

[Mark Johnson]

Trey Scarborough wrote:



- Original Message - From: Mark Johnson 
[EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 8:56 AM
Subject: [Asterisk-Users] Cisco 7750


I have read of people attempting to do this, and I just wanted 
everyone to know about what we've discovered about the Cisco 7750.  
If you don't know what it is, it's basically a blade server.  I have 
1 power blade, 1 alarm processor, 2 system processing engines and 1 
multi-service route processor. We just got asterisk running on this 
today!!!



Just dont let cisco know

We haven't tested the T1 with it, yet, but I pretty sure it will work 
OK. All of the FX ports work beautifully right now.  The big deal 
about this for me is that I  have battled over and over again with 
interrupt issues with Digium hardware.  This is sweet because all the 
T1 processing including echo cancellation should be done on the route 
processor. Asterisk doesn't have to do much of anything.




so im guessing that all of the t1/fx ports are configured in the 
system processor and just talk sip/mgcp to the route proccessor.


That sounds like a pretty sweet setup If you could only get cisco to 
sell you the hardware without having to buy the software.


I'm seeing that these things are on E-Bay pretty often.  They still want 
way too much money for what it is.  But if you where trying to get away 
from Call Manger and already owned one...


Mark
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Re: [Asterisk-Users] accountcode not present in cdr

[Max Clark]

But what do people do with large LCR rules... Build special contexts for 
each peer/user and then include the main LCR context? This seems a 
little cludgy.


Is there any way to have the dialplan context set the account for cdr 
based on the accountcode defined in the sip.conf? At least this way I 
could have a single, generic dialplan.


-Max

Andres wrote:



Max Clark wrote:


Hi all,

I have what I hope will be a simple problem. In my sip.conf I have 
defined the accountcode field (see below), and they system does not 
report any errors when I reload the configuration. However when I look 
at my cdr detail (either the csv on disk, or the mysql info) the 
accountcode that I have specified is missing. I have scoured the list 
and have seen a few postings on this with no solutions. What should I 
be looking at to debug this.


Thanks in advance,
Max



Setting the accountcode in sip.conf is totally unreliable.  It does not 
work in many cases.  Your best bet is to set it in a context via the 
command:

SetAccount([account]):  Set  the  channel account code for billing



I am running Asterisk 1.0.7 on CentOS 4.0:
Linux 2.6.9-11.EL #1 Wed Jun 8 16:59:52 CDT 2005 i686 i686 i386 GNU/Linux

[provider1]
accountcode=provider1
type=friend
host=10.1.1.1
dtmfmode=rfc2833
username=user
secret=12345
qualify=no
canreinvite=no
insecure=very
disallow=all
allow=ulaw
allow=gsm

[provider2]
type=peer
accountcode=provider2
secret=54321
username=user
host=10.1.1.10
dtmfmode=rfc2833






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[Asterisk-Users] [ot] wifi 3G gsm phone

[trixter http://www.0xdecafbad.com]

This phone runs symbian, has a built in camera for video conferencing,
blah blah blah.  Dunno yet if you have enough to make it a soft phone,
but odds are there is.  Could be another gsm alternative to also do voip
[InfoWorld: Top News] Motorola adds Wi-Fi to 3G phone for NTT DoCoMo
http://www.infoworld.com/cgi-bin/redirect?source=rssurl=http://www.infoworld.com/article/05/06/21/HNmotorola3gwifi_1.html

Thought the list members may want to know about this incase they were
researching converged devices (the ipaq 6xxx does similar stuff but
afaik isnt 3G)

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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RE: [Asterisk-Users] Extension Configuration Best Practice

[Jay Milk]

 Ok, so how are you guys coping with scenarios like this? 
 Managers working in the office during the day or mid day and 
 then in the afternoon, working remotely using their laptops? 

Give them two extensions and ring them both.  One's the hard-phone,
one's the soft-phone.

 |Rich is indeed correct, Asterisk does not yet support multiple
 |registrations for a single peer entry. Thus when you register 
 |the previous registration is discarded and the new one is 
 |used. Thus like he said, the last one that registered gets the call.

And asterisk will never do that, because that's not how SIP works.

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Re: SV: SV: [Asterisk-Users] Presence and IM?

[Michiel van Baak]

On 12:00, Tue 21 Jun 05, Anton Krall wrote:
 Can this hint system be used for gxp2000 phones or just for snoms? 
 

Right now the gxp2000 doesn't support it. I heard rumours on
this list that Grandstream is planning this feature for some
future firmware. I'm waiting for it as well. Till that time
I'll stick to snoms :)
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence.

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RE: [Asterisk-Users] Extension Configuration Best Practice

[Joshua Colp]

Actually SIP has the capability for it... For example, on Free World Dialup
which uses SER you can have up to 24 registered SIP devices to a single
account I believe, may be slightly smaller... But it's still a large number.
Thus when your number is rung, all registered SIP devices are contacted...
It's just that in the Asterisk world everything is designed with a one
device per peer concept.

- Joshua Colp. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, June 21, 2005 2:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Extension Configuration Best Practice

 Ok, so how are you guys coping with scenarios like this? 
 Managers working in the office during the day or mid day and then in 
 the afternoon, working remotely using their laptops?

Give them two extensions and ring them both.  One's the hard-phone, one's
the soft-phone.

 |Rich is indeed correct, Asterisk does not yet support multiple 
 |registrations for a single peer entry. Thus when you register the 
 |previous registration is discarded and the new one is used. Thus like 
 |he said, the last one that registered gets the call.

And asterisk will never do that, because that's not how SIP works.

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