Re: [Asterisk-Users] 2 problems

2005-11-03 Thread João Amaro 




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Viva,

Witch trunk are you able to establish ?

SIP trunk beetwen asterisk and PBX ?
Oh323 ?


Joo

Manuel Silva wrote:

| Hello!
|
| I have installed 2 servers, one with SER integrated with PostgreSQL
| (Fedora Core 3) and the other with Asterisk (Fedora Core 4). I can
| talk Softphone - SER - SER - Softphone (in case I try
to
| contact a person that as a different SIP server). Now the goal is
| to, use Asterisk as a gateway, with SIP trunking between the
| Asterisk and a PBX (with IP module).
|
| So if I try to make a call from a softphone to a PSTN connected
| phone, it will be like this:
| softphone-SER-Asterisk-PBX-destination.
|
| I have already configured SER to forward calls to Asterisk, and it
| seems to work fine. Now, my problem is how to configure Asterisk to
| forward a call from SER to the PBX, or how to establish an IP
| trunking between Asterisk and the PBX. The other problem is how do
| I configure Asterisk, to forward a call from a PSTN phone.
|
| If anyone can help me, please!! :-)
|
|
| Manuel Silva
|
|
|
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|
|
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[Asterisk-Users] Asterisk dies with Meetme

2005-06-28 Thread João Amaro 

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Hi List

I'm trying to create a conference room using H323 channels.

If i start asterisk normally (service asterisk restart) and connect to
cli using -vvvr options, when a user enters the Conference,
asterisk says You are the only ... and then dies, withou any error
message, nothing at all.
But, if i start asterisk with cli console (-vvc) theres is no
problem and i can create the conference rooms.

Zaptel and ztdummy modules are already loaded.
Anyone with the same problem ?

Asterisk: 1.0.8
Kernel: 2.6.9-5.0.3.ELsmp


Regards, João Amaro


 DUMP 
[EMAIL PROTECTED] ~]# service asterisk restart
[EMAIL PROTECTED] ~]# asterisk -r
(...)
~-- Executing MeetMe(OH323/R47, 1234|ciMps|) in new stack
~  == Parsing '/srv/etc/asterisk/meetme.conf': Found
2005-06-28 16:49:53 WARNING[4555]: channel.c:1913 ast_request: No
channel type registered for 'zap'
2005-06-28 16:49:53 WARNING[4555]: app_meetme.c:227 build_conf: Unable
to open pseudo channel - trying device
~-- Created MeetMe conference 1023 for conference '1234'
~-- Playing 'conf-onlyperson' (language 'en')
Vr-VoIP1*CLI
Disconnected from Asterisk server
Executing last minute cleanups

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Re: [Asterisk-Users] Using voicemail independently from Asterisk PBX

2005-04-20 Thread João Amaro 
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John Riek wrote:
| I would like to use Asterisk as a standalone voicemail server and
| integrate it with a Cisco Call Manager PBX. I need to know how to
| run the voicemail system independently.  Does anybody know how to
| do this?
|
What Version of CCM do you have ?
Is it CCM ou CCM Express ?
If it is CCM  4.0 then i advise you to use a SIP trunk between CCM and *.
Create your mailbox(s) in voicemail.conf, make your dialplan and route
the calls to * and it's done ;)
Rgds
João Amaro
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[Asterisk-Users] Urgent .... Asterisk - Cisco CCM SIP TRUNK

2005-04-15 Thread João Amaro 
Hi All,
I'm getting a strange problem with asterisk 1.0.6.
I've got a SIP Trunk between CCM 4.1 and Asterisk 1.0.6.  We are
talking about a daily average of 900 calls, and 600 minutes.
I'm running in asterisk 3 queues, 2 of them with dynamic members
(chan_local), and one with only one static member, and a Voicemail system.
Since yerstaday (i wonder why?), and only sometimes, when a call enter
a queue, it starts ringing a member phone, and when he tries to pickup
the call it jumps to other queue member, and when the other queue
member pick up the call, it jumps again to other queue member 
they are getting crazy   until one of thems catchs the call.
When it happens, in the next seconds if i try to dial to asterisk, CCM
gives-me a busy tone, but nothing reachs asterisk box, i dont now why
CCM's gives busy tone without sending the call to asterisk !!! Is this
a CCM feature ?
All the phones used are cisco (skinny) attached to CCM.
The only thing that appears on logs is:
   file.c:550 in ast_readaudio_callback: Failed to write frame.
Thanks in advance
Regards
  João Amaro

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Re: [Asterisk-Users] Asterisk .call files

2005-04-06 Thread João Amaro 




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Hi.

If you show what's in the .call file, it would help.

Rgs

Joao


Gilbert Abboud wrote:

| hi
|
| I created a .call file as mentioned in the WiKi but when i place it
| in /var/spool/asterisk/outgoing, the Asterisk console shows
| "unknown keyword" for all the keywords used in the .call file (i.e
| channel, context, extension,...). Any ideas why?
|
| Regards,
|
| Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst
| Excendia, Montreal ESN: 514-765-8490
|
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Re: [Asterisk-Users] Asterisk Voice mail with CCM

2005-04-03 Thread João Amaro 
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Nathan Reeves wrote:
| Anyone running Cisco Call Manager and using Asterisk for voice mail
|  services?  Things working well or is the concept a bit of a hassle
| to implement?
|
Hi,
I'm using asterisk with a SIP trunk as a voicemail system for CCM without
problems till now.
João Amaro
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Re: [Asterisk-Users] Call Queues and Transfers

2005-03-15 Thread João Amaro 




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Hi

How do you queue the incoming call ?
Do you queue the call with the t option (allow the called user to
transfer the calling user) ?

Regards

Joo Amaro

Anton Krall wrote:

| Guys.. Why is it that when a call comes to a call queue and in term
| gets assigned to an agent, if that agent tries to xfer the call
| using # or any other feature, it doesn't do anything? I just hear
| the "pleeps" on the phone but asterisk doesn't intervene with the
| "Transfer" prompt.
|
| Am I missing something?
|
| Thx!
|
|
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Re: [Asterisk-Users] speech recognition

2005-02-14 Thread João Amaro 




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Hi

David D. Faerman wrote:

| hi i am looking for some info for speech recognition for example
| when someone call to my house asterisk ask for who hi want to call
| and he say the name david or susan (wife) or daniela etc...
|
Why not the easy way ?

"Press 1 for Susan",
"Press 2 for David",
"Press 3 for Sam the Dog",
"Press 4 for Nemo the Little Fish",
"Press 5 to leave a message",
"Press 6 to Hangup".

rgds

Joo Amaro

| thanks David
|
|
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Re: [Asterisk-Users] ADM 0.5 - Asterisk Desktop Manager (alpha)

2005-02-07 Thread João Amaro 




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Hi Richard,

Your webserver:


~ Server Error: 403 Forbidden
-
--
~ Access denied
-
--
~ /doc/index.htm


Sds


Richard Hamnett wrote:

| Hi all,
|
| I've just released an ALPHA version of an application I have been
| working on to help integrate the desktop with asterisk. A list of
| the key features are as follows:
|
| * Automatic on-call volume reduction * One click dial from
| clipboard * Automatic on-call messenger away * Automatic/manual DND
| phone setting (Cisco 79xx only) * One click call forward setup
| (Cisco 79xx only)
|
| Bear in mind that some of these features are incomplete, but i'd
| like to get this out there, for people to test and find some bugs,
| I'd really appreciate any patches if you can iron out any bugs, and
| some feature suggestions if you have any. A full list of features
| (including intended ones) are listed on the ADM website:
|
| http://adm.hamnett.org
|
| Please test it out :)
|
| PS. This is primarily designed for Linux users, but I think it
| should work with Mac OSX 10.2/3 and Win32 (with gtk2 and gtk2-perl
| binaries installed) so if anyone could try to get it working with
| either of those O/S's that would be cool
|
| Regards Richard Hamnett
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[Asterisk-Users] OH323 Cisco Transfer Key

2005-01-25 Thread João Amaro 




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Hi All

I'm using * as a Call Center to CCM.
All the phones are ciscop ip phones (witk skinny) attached to CCM.

When i try to transfer a call, from one phone to another, when i press
the transfer key
i get this message on oh323.log:

~ [2]PAsteriskSoundChannel::Write: Write Failed (G.711) -
Destination Address Required

and i can't transfer the calls because the channels are broken.

However i can transfer the call using the # key (via asterisk), but i
want to know if is possible to do this using the
cisco transfer key (via ccm).


Thanks in advance


Joo Amaro



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Re: [Asterisk-Users] queue log analyser?

2005-01-20 Thread João Amaro 
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Ben Merrills wrote:
| There's a few (open source/free) ones in development. I myself am
| developing one of them.
|
| Ben
|
Hi.
Why not join all the project in just one ?
Actually which queue log analyzers projects are beeing developed ?
Check the mail from Ben Merrills sent to the list 14-10-2004 15:10.
I don't know if he releases the source code, but, from the screenshots
it seems to be a good one.
Jo?o Amaro
- -- Begin Mail
| I've been doing some work on a queue log analyser for a while now,
| getting the basics in place, an example of which you can find at
| the URL below. However, just wondering what information people
| think is most useful in a log analyser?
|
| At present it includes the following features:
|
| # Time periods - specify a period of days from the log which you
| want to generate statistics for (e.g. only the last 14 days) #
| Templating - allows the stats to be inserted into any html/text
| template using specific tags to insert stats. This means you could
| create a number of templates and execute the analyser against them
| to give different information on different pages (quite flexible).
| # Specify start and end dates - similar to the first feature,
| except you can specify a tight period from your log, not just the
| last x number of days # Channels/Agents to names - simple text file
| allows you to specify a name, agent number and a channel - e.g.
| Ben, Agent/1, Sip/ben. This is then used in the output # instead
| of raw data # JPG graphs - includes a custom class to generate line
| graphs of information (e.g. hourly call volumes etc)
|
| What I want to know though is, what output people would like. At
| the moment there is an overview of all queues, which includes:
|
| Total Calls, total connected calls, total abandoned calls, calls
| abandoned within x seconds, calls exited with key press, Average
| hold time, max hold time, average talk time
|
| Agent overview includes: Calls taken, Average talk time
|
| Graph of call volume per hour of the day Graph of call volume per
| day (over the period specified)
|
| Runs under windows (.NET or mono required) or any other OS that
| support .NET/mono (Linux, Mac, BSD etc)
|
| http://muad.xdev.net/Projects/qig/sample.html
|
|
| Not really done anything like this before, so as much input as
| possible would be appreciated.
|
| Cheers,
|
| Ben
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Re: [Asterisk-Users] Asterisk to PSTN

2005-01-10 Thread João Amaro 




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Hello

You can use H323 to connect to Cisco CallManager.
Add asterisk as an h323 gateway on cisco callmanager.
Then you can send  receive call from asterisk.

TIP: Use OH323 instead off asterisk h323 native driver.

Regards

Joo Amaro



Walid Azab wrote:

| I have installed [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] on a PC
here
| and need to have it forward calls to the PSTN. We have Cisco
| CallManager 3.3.4. However I found out that this version doesn't
| support configuring SIP Trunks. Is there an alternative solution.
| Thanks
|
| Walid
|
|
|
--
|
|
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Re: [Asterisk-Users] asterisk - oh323 driver

2005-01-06 Thread João Amaro 




Hi all


I've managed to get chan-oh323-0.6.5 working with asterisk-1.0.3
I've downloaded all the files from www.inaccessnetworks.com

  pwlib + pwlib-janus patch
  openh323 + openh323-janus patch
  chan-oh323 0.6.5

Don't forget to apply the chan-oh323 patch to openh323 before compiling.

Hope it helps

Regards

Joo Amaro


Kanuri, Seshu (Company IT) wrote:

  
  
  
  
  
  
  Joao wrote:
  
  
  Meanwhile i've downloaded ,again, the 0.6.5 version.
I'm using pwlib and openh323 versions from sourceforge.
It compiled without errors, but the error at startup it's the same
  
  This is the ldd output for the driver.
  
Shouldn't this be linked to the wrapper ?
  
  
I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4


I am trying to compile thge latest h323 libraries from openh323.org site 
and also from sourceforge and I get only one error as under:
/usr/include/ptlib/syncthrd.h:356: error: 'PDictionary' is used as a
type, but is not defined as a type.

The error seems to be in synthrd.h file at line 356, 
where it is used but not declared in the beginning. Does nyoneone know how to fix this?
My be we need to declare PDictionary as a type in the file. 
Does anyone know how to declare this in the header file?
Seshu Kanuri

  
  
  
  
  
  NOTICE: If
received in error, please destroy and notify sender. Sender
does not waive confidentiality or privilege, and use is prohibited.
  
  
  



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Re: [Asterisk-Users] asterisk - oh323 driver

2005-01-06 Thread João Amaro 




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Kanuri,


If you want to use the last stable release of asterisk (1.0.3), you
should do it:
(don't forget to read the README )

Get oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
Get pwlib from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz
Get asterisk-oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.6.5.tar.gz


Untar the files
#tar zxvf openh323-Janus_patch4-src-tar.gz
#tar zxvf pwlib-Janus_patch4-src-tar.gz
#tar zxvf asterisk-oh323-0.6.5.tar.gz
#tar zxvf asterisk-1.0.3.tar.gz


Install Pwlib
#cd pwlib
#./configure  make clean  make opt 
make install  ldconfig


Patch and Install OpenH323
#cd openh323
#patch -p1  ../asterisk-oh323-0.6.5/openh323_1.13.5-make.patch
#./configure  make clean  make opt 
make install  ldconfig


Asterisk
#cd asterisk-1.0.3
#make  make install  make samples


Asterisk-oh323
#cd asterisk-oh323-0.6.5
Edit the Makefile
#make  make install  ldconfig

Hope it helps,

Contact-me of line if it don't work


Joo Amaro







Kanuri, Seshu (Company IT) wrote:

|
| Rafael,
|
| Thanks for the detailed instructions. This really helps everyone
| looking fix this nagging issue.
|
| Seshu Kanuri
|
| -Original Message- From:
| [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED]] On Behalf Of
| Rafael J. Risco G.V. Sent: Thursday, January 06, 2005 9:45 AM To:
| Asterisk Users Mailing List - Non-Commercial Discussion Subject:
| Re: [Asterisk-Users] asterisk - oh323 driver
|
| Hi I am using oh323-0.7.1 with asterisk cvs head version and works
| great for me (Linux Fedora1), see details below:
|
| Requirements: PWLIB : pwlib-v1_6_6-src.tar.gz (or
Janus_Patch)
| OpenH323 : openh323-v1_13_5-src.tar.gz (or
Janus_Patch)
| Inaccessnetworks-asterisk-oh323 : asterisk-oh323-0.7.1.tar.gz
|
| Sources:
|
http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asteris
| k-oh323-0.7.1.tar.gz
|
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh3
| 23-Janus_patch4-src-tar.gz
|
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-
| Janus_patch4-src-tar.gz
|
| Note: asterisk-oh323-0.7.1 must be used with Asterisk CVS Head...
|
| Installation: tar -zxvf asterisk-oh323-0.7.1.tar.gz tar -zxvf
| pwlib-Janus_patch4-src-tar.gz tar -zxvf
| openh323-Janus_patch4-src-tar.gz
|
| cd pwlib ./configure make
|
| cd openh323 patch -p1 
| /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch (pach to
| openh323)
|
| cd openh323 ./configure make opt
|
| ASTERISK CVS Head: ---
|
| export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs
| login - the password is anoncvs. cvs checkout zaptel libpri
| asterisk
|
| cd zaptel/ make clean; make install
|
| cd libpri/ make clean make install
|
| cd asterisk/ make clean make install make samples make progdocs
|
| Finally install ASTERISK OH323 channel driver:
| -- cd
| asterisk-oh323-0.7.1 vi Makefile ( check paths according with your
| system )
|
| PWLIBDIR=/root/pwlib OPENH323DIR=/root/openh323
| ASTERISKINCDIR=/root/asterisk/include
| ASTERISKMODDIR=/usr/lib/asterisk/modules
| ASTERISKETCDIR=/etc/asterisk OH323WRAPLIBDIR=/usr/local/lib
| SSLINCDIR=/usr/include/openssl SSLLIBDIR=/usr/lib
|
| Compiling --- Type "make" to build the oh323wrap library
| and the ASTERISK OH323 channel driver.
|
| Type "make install" to install the binaries.
|
| Add to your LD_LIBRARY_PATH the path where the oh323wrap library
| was installed (or edit your /etc/ld.so.conf file, add the library
| path, and run "ldconfig").
|
| Hope it helps
|
| Rafael Risco
| 
|
| NOTICE: If received in error, please destroy and notify sender.
| Sender does not waive confidentiality or privilege, and use is
| prohibited.
|
|
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[Asterisk-Users] asterisk - oh323 driver

2005-01-05 Thread João Amaro 




Hi List


I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4

At starting time, i've this error message

# /srv/usr/sbin/asterisk -vvvc
[chan_oh323.so]
Jan  3 17:06:26 WARNING[5817]: loader.c:258 ast_load_resource: /srv/usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: _ZNK13PSoundChannel6IsOpenEv

Jan  3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module chan_oh323.so failed!


I've tried to upgrade to version 0.6.5, but i got a compile error. 
Anyone know how to solve this error ?

Thanks in advance, and have a GOOD 2005

Regardz,

Joo Amaro



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Re: [Asterisk-Users] asterisk - oh323 driver

2005-01-05 Thread João Amaro 




Hello,

Meanwhile i've downloaded ,again, the 0.6.5 version. I'm using pwlib
and openh323 versions from sourceforge.
It compiled without errors, but the error at startup it's the same

This is the ldd output for the driver.

Shouldn't this be linked to the wrapper ?

# ldd chan_oh323.so
 libstdc++.so.5 = /usr/lib/libstdc++.so.5 (0x009b3000)
 libpthread.so.0 = /lib/tls/libpthread.so.0 (0x00a69000)
 libldap.so.2 = /usr/lib/libldap.so.2 (0x00a79000)
 libldap_r.so.2 = /usr/lib/libldap_r.so.2 (0x00aa3000)
 liblber.so.2 = /usr/lib/liblber.so.2 (0x00efb000)
 libsasl.so.7 = /usr/lib/libsasl.so.7 (0x00c36000)
 libssl.so.4 = /lib/libssl.so.4 (0x00ad1000)
 libcrypto.so.4 = /lib/libcrypto.so.4 (0x00b05000)
 libexpat.so.0 = /usr/lib/libexpat.so.0 (0x00bf6000)
 libresolv.so.2 = /lib/libresolv.so.2 (0x00fa6000)
 libdl.so.2 = /lib/libdl.so.2 (0x00f0b000)
 libc.so.6 = /lib/tls/libc.so.6 (0x00c42000)
 libm.so.6 = /lib/tls/libm.so.6 (0x00d97000)
 libgcc_s.so.1 = /lib/libgcc_s.so.1 (0x00c16000)
 /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x00d81000)
 libgdbm.so.2 = /usr/lib/libgdbm.so.2 (0x00c1f000)
 libcrypt.so.1 = /lib/libcrypt.so.1 (0x00f17000)
 libpam.so.0 = /lib/libpam.so.0 (0x00c26000)
 libgssapi_krb5.so.2 = /usr/kerberos/lib/libgssapi_krb5.so.2
(0x00db9000)
 libkrb5.so.3 = /usr/kerberos/lib/libkrb5.so.3 (0x00dcc000)
 libcom_err.so.3 = /usr/kerberos/lib/libcom_err.so.3
(0x00c2e000)
 libk5crypto.so.3 = /usr/kerberos/lib/libk5crypto.so.3
(0x00f56000)
 libz.so.1 = /usr/lib/libz.so.1 (0x00e2a000)
 liblaus.so.1 = /lib/liblaus.so.1 (0x00c3)

Regards


Silviu Herchi wrote:

  Hi,

The key to this stuff is using the exact versions of the required libs and
following blindly the instructions (the pwlib and openh323 libraries from
sourceforge.net worked better in my case than the ones from
innaccessnetworks.com). What is the error message you get when you try to
compile asterisk-oh 0.6.5?

Regards,

Silviu


De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de Joo Amaro
Envoy: mercredi 5 janvier 2005 16:38
: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: [Asterisk-Users] asterisk - oh323 driver

Hi List


I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4


At starting time, i've this error message

# /srv/usr/sbin/asterisk -vvvc
[chan_oh323.so]
Jan  3 17:06:26 WARNING[5817]: loader.c:258 ast_load_resource:
/srv/usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK13PSoundChannel6IsOpenEv

Jan  3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module
chan_oh323.so failed!


I've tried to upgrade to version 0.6.5, but i got a compile error. 
Anyone know how to solve this error ?

Thanks in advance, and have a GOOD 2005

Regardz,

Joo Amaro

  



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Re: [Asterisk-Users] asterisk - oh323 driver

2005-01-05 Thread João Amaro 




Hello,

Meanwhile i've downloaded ,again, the 0.6.5 version. I'm using pwlib
and openh323 versions from sourceforge.
It compiled without errors, but the error at startup it's the same

This is the ldd output for the driver.

Shouldn't this be linked to the wrapper ?

# ldd chan_oh323.so
 libstdc++.so.5 = /usr/lib/libstdc++.so.5 (0x009b3000)
 libpthread.so.0 = /lib/tls/libpthread.so.0 (0x00a69000)
 libldap.so.2 = /usr/lib/libldap.so.2 (0x00a79000)
 libldap_r.so.2 = /usr/lib/libldap_r.so.2 (0x00aa3000)
 liblber.so.2 = /usr/lib/liblber.so.2 (0x00efb000)
 libsasl.so.7 = /usr/lib/libsasl.so.7 (0x00c36000)
 libssl.so.4 = /lib/libssl.so.4 (0x00ad1000)
 libcrypto.so.4 = /lib/libcrypto.so.4 (0x00b05000)
 libexpat.so.0 = /usr/lib/libexpat.so.0 (0x00bf6000)
 libresolv.so.2 = /lib/libresolv.so.2 (0x00fa6000)
 libdl.so.2 = /lib/libdl.so.2 (0x00f0b000)
 libc.so.6 = /lib/tls/libc.so.6 (0x00c42000)
 libm.so.6 = /lib/tls/libm.so.6 (0x00d97000)
 libgcc_s.so.1 = /lib/libgcc_s.so.1 (0x00c16000)
 /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x00d81000)
 libgdbm.so.2 = /usr/lib/libgdbm.so.2 (0x00c1f000)
 libcrypt.so.1 = /lib/libcrypt.so.1 (0x00f17000)
 libpam.so.0 = /lib/libpam.so.0 (0x00c26000)
 libgssapi_krb5.so.2 = /usr/kerberos/lib/libgssapi_krb5.so.2
(0x00db9000)
 libkrb5.so.3 = /usr/kerberos/lib/libkrb5.so.3 (0x00dcc000)
 libcom_err.so.3 = /usr/kerberos/lib/libcom_err.so.3
(0x00c2e000)
 libk5crypto.so.3 = /usr/kerberos/lib/libk5crypto.so.3
(0x00f56000)
 libz.so.1 = /usr/lib/libz.so.1 (0x00e2a000)
 liblaus.so.1 = /lib/liblaus.so.1 (0x00c3)

Regards


Silviu Herchi wrote:

  Hi,

The key to this stuff is using the exact versions of the required libs and
following blindly the instructions (the pwlib and openh323 libraries from
sourceforge.net worked better in my case than the ones from
innaccessnetworks.com). What is the error message you get when you try to
compile asterisk-oh 0.6.5?

Regards,

Silviu


De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de Joo Amaro
Envoy: mercredi 5 janvier 2005 16:38
: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: [Asterisk-Users] asterisk - oh323 driver

Hi List


I'm having problems starting asterisk  with asterisk-oh323-0.6.4.
I'm using this versions:
 asterisk-1.0.3
 asterisk-oh323-0.6.4
 openh323-Janus_patch4 + asterisk-0h323 patch
 pwlib-Janus_patch4


At starting time, i've this error message

# /srv/usr/sbin/asterisk -vvvc
[chan_oh323.so]
Jan  3 17:06:26 WARNING[5817]: loader.c:258 ast_load_resource:
/srv/usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK13PSoundChannel6IsOpenEv

Jan  3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module
chan_oh323.so failed!


I've tried to upgrade to version 0.6.5, but i got a compile error. 
Anyone know how to solve this error ?

Thanks in advance, and have a GOOD 2005

Regardz,

Joo Amaro

  



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Re: [Asterisk-Users] chan_oh323 Module for Asterisk

2005-01-05 Thread João Amaro 




That's the problem. 

You need the chan_oh323.so and the oh323wrapper.
You can try it, but, i guess it i'll not work.

A little help from Michael Manousos at this point i'll be great ;)

Tomorrow i'll try to get it working, but, if i can't, 
maybe i'll need to do downgrade asterisk  chan_oh323 versions.

Joo Amaro



Tenorio, Leandro wrote:

  I got it, but email it to the list is not a good option.

Who 're interested just email me, I'll send it asap.
But AFAIK, you still need the wrapper.

LTenorio
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Kanuri,
Seshu (Company IT)
Sent: Wednesday, January 05, 2005 5:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] chan_oh323 Module for Asterisk

 
If anyone in the list has a working version of the chan_oh323.so file
for Fedora Core 2 and Redhat, can he email the same to the list as
attachment. This will reduce the pain for many of the users who are
trying to compile the same from the libraries, which never seemed to
work.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Humberto
Aicardi
Sent: Wednesday, January 05, 2005 3:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] chan_oh323  gatekeeper

Hi folks,

	Until now I have used only SIP  IAX2 with success and
understand them pretty well. The point is that someone has asked me to
configure an * box for them, the problem is that they want to use H.323.
I have already compiled and tested the chan_oh323 with asterisk and
works. The problem is that the tests need a gatekeeper, my question is:
Do I need always need a gatekeeper? Or my FXO H.323 gateway can register
with * ?

Thanks,
Humberto Aicardi

 
NOTICE: If received in error, please destroy and notify sender.  Sender
does not waive confidentiality or privilege, and use is prohibited. 
 
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Re: [Asterisk-Users] Best Linux base for small Asterisk server?

2004-11-04 Thread João Amaro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
I'm using Fedora RC1 without problems (kernel 2.4.22)
Anyone here have tried Whitebox Respin 1 with asterisk ?
Maybe nest week i'll install a small Asterisk Server on it.
Asterisk + Apache only.



Bill Bradford wrote:
| I'm in the process of building up a small (1x1) test Asterisk box
| based on a 1Ghz VIA C3 Mini-ITX box with one PCI slot (and a
| FX100P).
|
| Anyone have suggestions as to the best Linux distribution (or
| kernel) to base the system on?
|
| I'll just have one FXO/POTS line and then a Grandstream Budgetone
| 101 IP phone; this is more for playing with IVR functionality than
| anything else.
|
| Thanks.
|
| Bill ___ Asterisk-Users
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|
|
- --
Manuel João S. Costa Amaro [EMAIL PROTECTED]
ICQ: 57398499
MSN: [EMAIL PROTECTED]
As únicas pessoas que aprecio são os loucos: os que são loucos para
viver, loucos para falar, loucos para se salvar, desejosos de tudo ao
mesmo tempo; os que nunca bocejam nem dizem lugares-comuns, mas que
ardem, se inflamam e brilham como fabulosos fogos-de-artifício. 
(Jack Kerouac)

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Re: [Asterisk-Users] How useful is the screen on IP phones?

2004-10-25 Thread João Amaro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Paul Crick wrote:
| Hey Jay
|
| All the stuff you've described is possible. I've done some playing
| with the XML services on a Cisco 7960 to give ACD queue stats and
| system uptime info. The phone has a mini web browser built in so
| it's pretty easy to knock up some glue scripts in the back end to
| do what you want to do. I think there are some examples on the wiki
| too.
|
| Cheers Paul
|
Hi
I'm using 14 cisco 7940 as Dynamic queue agents.
They use the pixel based screen, to login/logout from queues. They can
also see the queues stats.

| ___ Asterisk-Users
| mailing list [EMAIL PROTECTED]
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|
|
- --
Manuel João S. Costa Amaro [EMAIL PROTECTED]
ICQ: 57398499
MSN: [EMAIL PROTECTED]
As únicas pessoas que aprecio são os loucos: os que são loucos para
viver, loucos para falar, loucos para se salvar, desejosos de tudo ao
mesmo tempo; os que nunca bocejam nem dizem lugares-comuns, mas que
ardem, se inflamam e brilham como fabulosos fogos-de-artifício. 
(Jack Kerouac)

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Re: [Asterisk-Users] How useful is the screen on IP phones?

2004-10-25 Thread João Amaro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Kevin P. Fleming wrote:
| João Amaro wrote:
|
| I'm using 14 cisco 7940 as Dynamic queue agents.
|
| They use the pixel based screen, to login/logout from queues.
| They can also see the queues stats.
|
|
| Now that's really not fair, to post a message like this without
| links to the code and/or documentation :-(
|
| Can we assume from your message that you have implemented this
| yourself but are not making it available to the community?
|
|
Hi
It's not finished yet.
I've to make some changes, because right now i've made it to work with
my configuration (with just 2 queues).
It's based on qview.pl from contrib files in asterisk source.
I'll share it :)
You can contact me by email .


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[Asterisk-Users] Queues Problems

2004-10-21 Thread João Amaro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
I'm using asterisk (1.0-RC2) as a callcenter, with IVR, for cisco
callmanager (via H323).
When the caller calls the callcenter, he will be prompted to press 1
or 2. For  QUEUE1 and QUEU2.
While waiting i've musiconhold playing . All the agents are dynamic.
I'm using the roundrobin algorithm.
Recently, i'm having some problems. Sometimes, when an anget pickups
the call, he starts talking with the caller, and everything is OK ...
but, and i dont know why, the caller goes back to MusicOnHold: the
caller keeps waiting (moh) while the call is picked up by other queue
agent.
Strange ...
Regards
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[Asterisk-Users] MusicOnHold and Mp3 threads

2004-09-23 Thread João Amaro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
I putting two * boxes  into production. It's a callcenter + voicemail
to Cisco callmanager.
My problem is that mpeg123 sometimes doesn't terminate.
What should i do ? Don't use MusincOnHold, and use a single MP3 file
with a high length ?

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[Asterisk-Users] RC2 with OH323 or H323

2004-09-03 Thread João Amaro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
I've just finished my upgrade to asterisk RC2.
I need to have H323 support, and in the last months i've been using
the chan-oh323 with good results.
My question is: anyone in the list have made tests with both chans
(oh323 and h323), which is best ?
For this installation i don't need the gatekeeper support, i just want
to receive/place calls to Cisco CallManager.
If anyone tried to install OH323 with asterisk RC2 with success,
please send-me an email. I can compile the driver and the
library, but i can't initialize the driver when i start the asterisk.

Thanks ind advance,
Regards,
João Amaro
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[Asterisk-Users] Agents Log off

2004-09-01 Thread João Amaro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
I'm using the apllication AgentCallBackLogin so agents can login to a
queue. They just need to
enter the password, the CallBack Extensions is the ${CALLERIDNUM}
Is there a way to AgentsLogOff withou using the AgentCallBackLogin
application. I don't want the
user to enter they CALLERIDNUM.
Regards

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