[asterisk-users] Linksys SPA941
Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing the Caller ID
Dear Group,
I have a scenario where I would like to change the caller ID based on
the number dialled;
For example;
;Outbound UK and London Calls
exten=_8.,1,Set(CALLERIDNAME=0207100)
exten=_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten=_80039.,1,Set(CALLERIDNAME=0039024070)
exten=_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten=_80034.,1,Set(CALLERIDNAME=003491187)
exten=_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten=_80049.,1,Set(CALLERIDNAME=0049891214)
exten=_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
So for anything outside the Madrid, Milan, Munich extensions I would
like to use the generic UK number or have a pieced of logic that goes;
exten=_800.,1,Set(CALLERIDNAME=440207100)
exten=_800.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
Unfortunately this does not work, every time I dial I get
CALLERIDNAME=0207100.
What am I missing?
Many Thanks
Shad Mortazavi
n|m Nexus Management
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[asterisk-users] Mobile SIP Client
Dear All, I'm looking for a mobile SIP client to use with Asterisk. Has anyone got experience in this area and can you advise me of a product? Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management plc SIP: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mobile SIP Client
Thank you for the information. I'm specifically looking for a Windows 5.0 Mobile SIP agent for a Qtek 9000. Many Thanks Shad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem With Transfering Calls.
Dear Group, I have a requirement for the agents in the call queue to be able to transfer calls to other people within the organization and/or outside. Unfortunately when I add tT to the Dial Command i.e. exten = 0423,1,Dial(SIP/phone51,20,tT) When the agent presses # to acknowledge the call it sometimes starts the transfer process and does not acknowledge the call. Can someone please explain what the issue is here and how I can overcome this? Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management plc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatic 3 Way Call
Dear Group, I'm working on a call recording solution and would like to have the ability to initiate a 3 way call based on an incoming call. One party will be an AGI that I have other will be an outbound call via a second T1 interface. Does anyone have a working configuration for an Asterisk initiated 3 way call? Thanks and Regards Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc SIP: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatic 3 Way Call
Dear Group, I'm working on a call recording solution and would like to have the ability to initiate a 3 way call based on an incoming call. One party will be an AGI that I have other will be an outbound call via a second T1 interface. Does anyone have a working configuration for an Asterisk initiated 3 way call? Thanks and Regards Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc SIP: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New SkypeSIP gateway
Message: 24 Date: Mon, 03 Apr 2006 19:21:57 -0500 From: Michael Graves [EMAIL PROTECTED] Subject: [Asterisk-Users] New SkypeSIP gateway To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Anyone seen or tried this yet? http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php Michael - I have tried to register with both Asterisk and SER; Unfortunately this does not seem to work. Great idea. Guess we need to wait for the next version. I'll post some comments to the nch website. Shad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group,
I was able to fix this problem;
The solution was to use a prefix to dial out.
The next challenge was to send the SIP Domain over IAX2!. I found that
if I included @SIPDOMAIN it would break the IAX2 communications.
exten = _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/[EMAIL PROTECTED]),
breakes because @SIPDOMAIN is treated as the target context. You also
can not include @Context after the @SIPDOMAIN.
I created a new variable DS which was a concatenation of EXTEN and
SIPDOMAIN separated by % and not @ and I was now able to pass this over
IAX2;
DS = EXTEN%SIPDOMAIN.
exten = _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${DS}).
At the other end I used the CUT command and substring facilities in
Asterisk to split DS by the % eliminator; I re-formed a new variable
which was
DS = [EMAIL PROTECTED]
I can now pass calls from my internal Asterisk server to my external
Asterisk server using IAX2 and then call any external VoIP number.
Warm Regards
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
-Original Message-
From: Shad Mortazavi
Sent: Thursday, March 30, 2006 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten = shad,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
This answers part of the question;
However what I want to do is to send any outbound sip calls via our
external SIP server.
i.e;
VPN LANIAX2DMZ Internet
Internal UA --- Internal (*) -- External (*)--
ExternalUA
We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc.
Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?
Thanks
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
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[Asterisk-Users] How do you perform a Variable Substitution In Asterisk
Dear Group;
I have a requirement to pass the ${SIPDOMAIN} variable from Server A to
Server B over IAX2. Basically Server A is an Internal (*) and Server B
is an External (*) in the DMZ.
On Server A I do the following;
[SIPOUT]
exten = _6.,1,SetVar(DS=${EXTEN}%${SIPDOMAIN})
exten = _6.,2,Dial(IAX2/bxx:[EMAIL PROTECTED]/${DS})
On the CLI I get;
-- Executing Dial(SIP/phone6-bd3d,
IAX2//bxx:[EMAIL PROTECTED] /6shad%xxx..com) in new stack
This comes through over IAX2 and I can strip the 6 and send the call out
via SIP to my SIP proxy.
The only item missing is to substitute the % with @. Can this be done
natively in Asterisk? My production version is Asterisk CVS-v1-0-07.
I have read through
http://www.voip-info.org/wiki/view/Asterisk+variables and could see no
obvious method for this.
Many Thanks
Shad Mortazavi
-
Nexus Group Technical Manager
n|m Nexus Management Inc
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Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten = shad,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
This answers part of the question;
However what I want to do is to send any outbound sip calls via our
external SIP server.
i.e;
VPN LANIAX2DMZ Internet
Internal UA --- Internal (*) -- External (*)--
ExternalUA
We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc.
Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?
Thanks
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
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RE: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I am closer to where I want to be. I could still do with some help.
For my Internal(*)I setup the following;
extensions.conf
---
[SIPOUT]
exten = _6.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
If I dial sip:[EMAIL PROTECTED] I see the call go to the External(*)
In my external server I have;
Sip.conf
-
[sip_proxy-out]
type=peer ; we only want to call out, not be called
secret=
username=nexus*** ; Authentication user for outbound
proxies
fromuser=nexus*** ; Many SIP providers require this!
fromdomain=.***.com
host=
usereqphone=yes
and in the extensions.conf I have;
exten =_6.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
This all works!
The problem is it only works if I dial a user that exists on the SER
Server. eg sip:[EMAIL PROTECTED] .
It breaks if I call [EMAIL PROTECTED]
When I look at the INVITE packets the URI is being transformed when it
goes from the Internal(*) to the external (*) over IAX2. Rather than
being [EMAIL PROTECTED] it is translated to [EMAIL PROTECTED] !
This explains why calls to users on the SER server work.
I would appreciate an explanation of this phenomena and how to preserver
my URI going form the internal(*) to the external(*).
Warm Regards and Thanks
Shad Mortazavi
---
Nexus Group Technical Manager
n|m Nexus Management Inc
-Original Message-
From: Shad Mortazavi
Sent: Thursday, March 30, 2006 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten = shad,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
This answers part of the question;
However what I want to do is to send any outbound sip calls via our
external SIP server.
i.e;
VPN LANIAX2DMZ Internet
Internal UA --- Internal (*) -- External (*)--
ExternalUA
We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc.
Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?
Thanks
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
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[Asterisk-Users] Routing SIP calls via URI
Dear All,
I have the following setup;
SER/External Asterisk -- Firwall -- Internal Asterisk -VPN- Users
At the moment;
Anybody can register with our SER proxy and call each other using VoIP.
Anybody can call one of our internal users via our SER/Asterisk gateway.
The INVITE is sent to our external Asterisk Server, this act as a UA and
uses IAX2 to send the call to our internal Asterisk server.
Our internal users use a VPN to connect to our corporate HQ. They
register with our Internal Asterisk server and can make internal and
PSTN calls.
What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:[EMAIL PROTECTED] I would
like the call to be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk server
to act as a UA and make the call.
I have tried the following syntax on our internal server;
exten = _sip.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
However this does not seem to work?
How do I change my dial plan so that SIP calls are routed from my
internal Asterisk box to my external Asterisk box over IAX2?
Warm Regards and Thanks
Shad Mortazavi
---
Nexus Management Inc
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[Asterisk-Users] Possible AGI Bug in Asterisk?
Dear All,
I seem to have stumbled across an AGI problem;
I have written an AGI Script (bottom of this email);
The script does the following;
Makes a CDR entry when called
Records the call
Updates the CDR
Finds a corresponding DNIS from the SMDR table (captured via a serial
port logger)
Matches up the record and updates the CDR.
The script works perfectly in my test lab and has been doing so for
months. I have moved the script over to the production environment and
the script stops after;
$AGI-record_file($fname,'gsm',9,-1);...
Since the test environment was different to the production environment;
I upgraded my test environment from Asterisk V1 and RedHat 9 to V1.2.4
and CentOS.
Now in the test environment my script stops after;
$dbh -do(Update CDR SET Recording='0',GSM_File_Size='$size' WHERE
GSM_File = '$finame');
In both instances there are no perl errors and no errors reported in the
CLI. In fact the CLI comes back and tells me everything is OK.
Is this a known issue? Has someone else run into this? Is there a fix or
patch?
It looks to me like some type of timing issue with the AGI?
Any help would be appreciated. I have a production deadline.
Warm Regards
Shad Mortazavi
#!/usr/bin/perl
# CCVR Recording Module Shad Mortazavi December 15, 2005
use Asterisk::AGI;
use DBI;
#Create a DB-DNS
my $dbh = DBI
-connect('DBI:mysql:CCVR;192.168.6.56','CCVR_User','');
my $dbh1 = DBI
-connect('DBI:mysql:CCVR_ADMIN;192.168.6.56','CCVR_User','');
my $dbh3 = DBI -connect('DBI:mysql:CCVR;172.16.1.233','user1','');
#Create AGI
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
# Capture the extension dialed
my $extn = $input {'extension'};
#Answer the Call
$AGI-answer();
#generate a file name based on the time and extension dialed
$ti = time();
$dir = /vol/recordings/;
$finame = $ti._.$extn;
$fname = $dir.$finame;
$finame = $finame..gsm;
# populate the DB with the required information
# Agent, Filename, date, time
# Get the Date and Time
$gdate = $dbh-selectrow_array (SELECT current_date());
$gtime = $dbh-selectrow_array (SELECT current_time());
$date_time = $gdate $gtime;
# Cross reference the Agent Phone with extension
$phone = $dbh1-selectrow_array (SELECT phone from map where extn =
'$extn');
$agent = $dbh1-selectrow_array (SELECT UserID from agent where phone =
'$phone');
# Put data in DB
$dbh -do(INSERT INTO CDR
(recording_date,Extension,Phone,Agent,GSM_File,GSM_File_Size,State,Recor
ding,Telco_DNIS,Assigned_DNIS,Campaign) VALUES
('$date_time','$extn','$phone','$agent','$finame','$size','0','1',''
,'',''));
# Log Transaction
$dbh1 -do(INSERT INTO agent_log (date, action,agent, note) VALUES
('$date_time','Recording','$loginname','Agent $agent Recording $GSM_File
on phone $phone'));
# Start Recording
$AGI-record_file($fname,'gsm',9,-1);
# Get the file size
$fname = $fname..gsm;
($dev,$ino,$mode,$nlink,$uid,$gid,$rdev,$size,
$atime,$mtime,$ctime,$blksize,$blocks) = stat($fname);
# Put data in DB
$dbh -do(Update CDR SET Recording='0',GSM_File_Size='$size' WHERE
GSM_File = '$finame');
### This code is for people using the serial_logger to extract
information from a PBX
$agi_sleep_time = $dbh1-selectrow_array (SELECT agi_sleep_time from
system_variables where id = '1');
$SMDR_log_Variance = $dbh1-selectrow_array (SELECT SMDR_log_Variance
from system_variables where id = '1');
# Sleep to make sure we get the serial input we need following the call
sleep $agi_sleep_time;
# Get the date from the SMDR table
# Get the Date and Time
$gdate = $dbh-selectrow_array (SELECT current_date());
$gtime = $dbh-selectrow_array (SELECT current_time());
$date_time = $gdate $gtime;
# Get information from SMDR DB
$Telco_DNIS = $dbh3-selectrow_array (SELECT DNIS from SMDR where Extn
= '$phone' and CCVR_Date_Time = ('$date_time' - INTERVAL
$SMDR_log_Variance SECOND));
$Digits = $dbh3-selectrow_array (SELECT Digits from SMDR where Extn =
'$extn' and CCVR_Date_Time = ('$date_time' - INTERVAL
$SMDR_log_Variance SECOND));
if ($Telco_DNIS){
# Get information from DNIS Table
$Campaign_ID = $dbh1-selectrow_array (SELECT Campaign_ID from DNIS
where Telco_DNIS = '$Telco_DNIS');
$Assigned_DNIS = $dbh1-selectrow_array (SELECT DNIS from DNIS where
Telco_DNIS = '$Telco_DNIS');
$Campaign = $dbh1-selectrow_array (SELECT Campaign from campaign where
id = '$Campaign_ID');
}else{
$Telco_DNIS = ;
$Campaign = ;
$Assigned_DNIS = ;
}
# Update the call record
$dbh -do(Update CDR SET Telco_DNIS ='$Telco_DNIS', Assigned_DNIS
='$Assigned_DNIS', Campaign = '$Campaign', DDID = '$Digit' WHERE
GSM_File = '$finame');
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[Asterisk-Users] RE: Possible AGI Bug in Asterisk?
Dear All,
As a quick update. Switching back to Asterisk CVS-v1-0-02 seems to have
fixed the problem.
In addition ...
The command..
print STDERR Bla Bla Bla :\n;
Seems also to be broken in the new version see below;
New Version..
-Original Message-
From: Shad Mortazavi
Sent: Wednesday, February 08, 2006 11:28 AM
To: asterisk-users@lists.digium.com
Subject: Possible AGI Bug in Asterisk?
Dear All,
I seem to have stumbled across an AGI problem;
I have written an AGI Script (bottom of this email);
The script does the following;
Makes a CDR entry when called
Records the call
Updates the CDR
Finds a corresponding DNIS from the SMDR table (captured via a serial
port logger)
Matches up the record and updates the CDR.
The script works perfectly in my test lab and has been doing so for
months. I have moved the script over to the production environment and
the script stops after;
$AGI-record_file($fname,'gsm',9,-1);...
Since the test environment was different to the production environment;
I upgraded my test environment from Asterisk V1 and RedHat 9 to V1.2.4
and CentOS.
Now in the test environment my script stops after;
$dbh -do(Update CDR SET Recording='0',GSM_File_Size='$size' WHERE
GSM_File = '$finame');
In both instances there are no perl errors and no errors reported in the
CLI. In fact the CLI comes back and tells me everything is OK.
Is this a known issue? Has someone else run into this? Is there a fix or
patch?
It looks to me like some type of timing issue with the AGI?
Any help would be appreciated. I have a production deadline.
Warm Regards
Shad Mortazavi
#!/usr/bin/perl
# CCVR Recording Module Shad Mortazavi December 15, 2005
use Asterisk::AGI;
use DBI;
#Create a DB-DNS
my $dbh = DBI
-connect('DBI:mysql:CCVR;192.168.6.56','CCVR_User','');
my $dbh1 = DBI
-connect('DBI:mysql:CCVR_ADMIN;192.168.6.56','CCVR_User','');
my $dbh3 = DBI -connect('DBI:mysql:CCVR;172.16.1.233','user1','');
#Create AGI
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
# Capture the extension dialed
my $extn = $input {'extension'};
#Answer the Call
$AGI-answer();
#generate a file name based on the time and extension dialed
$ti = time();
$dir = /vol/recordings/;
$finame = $ti._.$extn;
$fname = $dir.$finame;
$finame = $finame..gsm;
# populate the DB with the required information
# Agent, Filename, date, time
# Get the Date and Time
$gdate = $dbh-selectrow_array (SELECT current_date());
$gtime = $dbh-selectrow_array (SELECT current_time());
$date_time = $gdate $gtime;
# Cross reference the Agent Phone with extension
$phone = $dbh1-selectrow_array (SELECT phone from map where extn =
'$extn');
$agent = $dbh1-selectrow_array (SELECT UserID from agent where phone =
'$phone');
# Put data in DB
$dbh -do(INSERT INTO CDR
(recording_date,Extension,Phone,Agent,GSM_File,GSM_File_Size,State,Recor
ding,Telco_DNIS,Assigned_DNIS,Campaign) VALUES
('$date_time','$extn','$phone','$agent','$finame','$size','0','1',''
,'',''));
# Log Transaction
$dbh1 -do(INSERT INTO agent_log (date, action,agent, note) VALUES
('$date_time','Recording','$loginname','Agent $agent Recording $GSM_File
on phone $phone'));
# Start Recording
$AGI-record_file($fname,'gsm',9,-1);
# Get the file size
$fname = $fname..gsm;
($dev,$ino,$mode,$nlink,$uid,$gid,$rdev,$size,
$atime,$mtime,$ctime,$blksize,$blocks) = stat($fname);
# Put data in DB
$dbh -do(Update CDR SET Recording='0',GSM_File_Size='$size' WHERE
GSM_File = '$finame');
### This code is for people using the serial_logger to extract
information from a PBX
$agi_sleep_time = $dbh1-selectrow_array (SELECT agi_sleep_time from
system_variables where id = '1');
$SMDR_log_Variance = $dbh1-selectrow_array (SELECT SMDR_log_Variance
from system_variables where id = '1');
# Sleep to make sure we get the serial input we need following the call
sleep $agi_sleep_time;
# Get the date from the SMDR table
# Get the Date and Time
$gdate = $dbh-selectrow_array (SELECT current_date());
$gtime = $dbh-selectrow_array (SELECT current_time());
$date_time = $gdate $gtime;
# Get information from SMDR DB
$Telco_DNIS = $dbh3-selectrow_array (SELECT DNIS from SMDR where Extn
= '$phone' and CCVR_Date_Time = ('$date_time' - INTERVAL
$SMDR_log_Variance SECOND));
$Digits = $dbh3-selectrow_array (SELECT Digits from SMDR where Extn =
'$extn' and CCVR_Date_Time = ('$date_time' - INTERVAL
$SMDR_log_Variance SECOND));
if ($Telco_DNIS){
# Get information from DNIS Table
$Campaign_ID = $dbh1-selectrow_array (SELECT Campaign_ID from DNIS
where Telco_DNIS = '$Telco_DNIS');
$Assigned_DNIS = $dbh1-selectrow_array (SELECT DNIS from DNIS where
Telco_DNIS = '$Telco_DNIS');
$Campaign = $dbh1-selectrow_array (SELECT Campaign from campaign where
id = '$Campaign_ID');
}else{
$Telco_DNIS
[Asterisk-Users] Asterisk and Inter-tel
Dear All, I'm in the process of writing a voice recording application for a customer using an Inter-Tel PBX. I was planning on using the T1 interface on the Inter-Tel to route calls to an Asterisk server that would do the voice recording. I was going to use the Caller ID or DNIS to help with some of the call details. I was told by the customers PBX vendor that the T1 on the Inter-tel could not be programmed to pass DNIS and/or caller ID to my Asterisk server over the T1 and that this function was only possible via the OAI interface?! Does anyone have anything like this working over a T1? Can someone confirm whether or not a T1 from an Inter-tel platform will carry DNIS and/or caller ID.? Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Hold Music is breaking up
Dear All, I have installed a Mediatrix 1204 on a client site and I'm sending calls between my Asterisk Server and the clients PBX over a VPN. The call quality is very good when I'm speaking with the staff and there are no breakups. The only problem I'm running into is the hold music is 'choppy'. If I call in over the T1 or using my Softphone there are no such problems. I was wondering if anyone could point me in the right direction. Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Voicemail while in queue.
Dear Group, I have the following requirement; I would like our users to be able to press 0 while they are in a call queue and have the option of leaving a voicemail, also when nobody is logged in, drop directly to a voicemail box. Is this possible? Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Outbound Mediatrix 1204.
Dear Group,
I have been able to configure my Asterisk BOX to receive calls from
Mediatrix 1204.
I'm having problems sending calls out via my Mediatrix unit.
The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends
back a Status : 480 Temporarily Unavailable.
This is my configuration on Asterisk;
exten = _78996.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _78996,1,Congestion
On the Mediarix end I have defined my SIP proxy.
I had to enable automatic call direction on my 1204 to send the call to
my Asterisk. If I leave this enabled and make an outgoing call via the
Asterisk box the call come back!
I have seen that several people have this feature working and would be
very grateful if you could share your configuration with me or point me
in the correct direction.
Thanks and Regards
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
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[Asterisk-Users] Mediatrix 1204 and Asterisk
Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
Dear All, I have been running an Asterisk 0.7.1 (patched with various agent applications) server for almost 2 years. We have a data center in the USA and a call center in the UK. All calls are routed to a group of central call queues in the USA. Agents from the data center, call center and from remote locations (London, Scotland, LA, Florida, and Maine) can log in, join the call queue and pick up calls This function has worked well since implementing the system and works well using SNOM 200's (data center and call center) and SJ Phone Build 1.50.271d, Mar 11 2005. I have rebuilt an identical test environment in my test lab and I can run version 0.7.1 (patched). I log in as an agent using my softphone, make a call from a second phone, I get greeted, put in a queue, given my position, the call goes through to my soft phone, I accept the call, press # and I'm on the call. I run the upgrade to version 1.0.9 and run the same test; I get greeted, put in a queue, given my position, the call goes through to my soft phone, and I accept the call, press #... I then get a message telling me that the system saying transfer? I see nothing on the CLI except the usual waiting for '#' to acknowledge To discount the SJ Phone I installed the version of X-Ten light that some of our agents/staff use and I got the same result. I checked the DTMF setting in sio.conf and these appear correct. I downgrade to 0.7.1 and the function works on both SJ Phone and X-ten light. I have included the CLI captures below; Sip show agents; (Angela Holt) available at '[EMAIL PROTECTED]' (musiconhold is 'default') From the CLI -- outgoing agentcall, to agent '1031', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1031 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/phone6SIP/0401|20|tr) in new stack -- Called phone6 -- Called 0401 -- Agent/1031 is ringing -- SIP/phone6-1d2b is ringing -- Agent/1031 is ringing -- SIP/phone6-1d2b answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge I need to move to the latest version of Asterisk to enable me to measure the number of minutes a user has been held in a queue. This function was not available in version 0.7.1. I remember a similar problem with version 0.7.2. Anyone else run into the same issue? Is it a known issue/bug? What is the fix? Thanks and Regards Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ 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
[Asterisk-Users] RE: Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
Good day Adam, I have about 30 Queues configured so at the risk of boring everyone I have included one of the lines; exten = _812108,1,Playback(nexus/wel-helpdesk-interwise) exten = _812108,2,SetCIDName(Client1) exten = _812108,3,Queue(Client1|Tt|||) exten = _812108,4,Playback(nexus/im-sorry) exten = _812108,5,Voicemail(1500) The _812108 is the DNIS number on the T1. I did have Tt configured in the queue. I followed your suggestion and changed this to; exten = _812108,1,Playback(nexus/wel-helpdesk-interwise) exten = _812108,2,SetCIDName(Client1) exten = _812108,3,Queue(Client1) exten = _812108,4,Playback(nexus/im-sorry) exten = _812108,5,Voicemail(1500) Same issue. I looked at the Agent's extension. It was configured as; ; Angela Holt exten = 0420,1,Dial(SIP/phone21,20,tr) exten = 0420,2,VoiceMail,u1021 exten = 0420,3,MusicOnHold(default) I changed this to; ; Angela Holt exten = 0420,1,Dial(SIP/phone21,20) exten = 0420,2,VoiceMail,u1021 exten = 0420,3,MusicOnHold(default) Removing the tr has done the trick. And the problem is gone. The agent can still transfer the call. Thanks for the idea. Warm Regards Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc I get greeted, put in a queue, given my position, the call goes through to my soft phone, and I accept the call, press #... I then get a message telling me that the system saying transfer? I see nothing on the CLI except the usual waiting for '#' to acknowledge Send the complete extensions.conf for the incoming call portion, and the agentcallbacklogin section. Also send the complete CLI from the call arriving into the PABX through to the call being sent to the agent. I suspect somewhere you are including the t or T option to the queue or dial which allows # to transfer a call. Of course, perhaps someone should check this, as we can't transfer a call until after we accept it... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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
[Asterisk-Users] I don't get a queue_log with my version of asterisk (0.7.1).
Dear All, I would like to add this feature to my version of asterisk. Is there a patch I can apply to get this function? Does anyone have any instructions for this? Thanks Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ 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
[Asterisk-Users] Calculating the lenght of time in a call queue?
Dear All, I'm running version 0.7.1 of Asterisk server for our global help desk. We have put together a comprehensive reporting package for static's from the CDR. I'm not able to calculate the time a call is in the queue before it goes to an agent? I would appreciate help with working this out. Warm Regards and Thanks Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ 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
[Asterisk-Users] RE:Calculating the lenght of time in a call queue?
I don't get a queue_log file? At what stage was this introduced? Thanks Shad ___ 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
[Asterisk-Users] Asterisk queue_log
Title: Asterisk queue_log Dear All, I'm running asterisk 0.7.1 on a production call center and second call center on Asterisk CVS-HEAD-05/18/04-01:57:42. On the Asterisk CVS-HEAD-05/18/04-01:57:42 box I have a queue_log file that I can use to get statistics. I get no such file in 0.7.1. Is this release dependent or is there configuration file I'm missing. Thanks and Regards Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ 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
[Asterisk-Users] SIP/SIMPLE, Jabber and Asterisk
Title: SIP/SIMPLE, Jabber and Asterisk Dear All, Is there an implementation of SIP/Simple for Asterisks? It would be neat to tie Asterisk to an IM like Jabber for presence. I believe this is already available for SER. Can anyone tell me if this is on the roadmap? I have been using both Asterisk and Jabber for quiet some time and would love to see these two working with each other. Would welcome any input on this. Shad Mortazavi Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Another) Queue log analyser
Title: (Another) Queue log analyser Ben, I would definitely have use for this application, fantastic start. When will you be making the source available? In my reports I use the CLID to look at calls for different agents i.e. call volume by agent. Warm Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney Message: 4 Date: Fri, 15 Oct 2004 09:33:26 +0100 From: Ben Merrills [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] (Another) Queue log analyser To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi there, Cheers for your suggestions, would be great to see the output of some other reports. Logins and logouts are available within the engine, just need to represent them in some way now. What do you suggest would be a good format? Typical duration of login? Only problem might be where someone hasn't logged out before their next login statement (no one here ever logs out, because they're all to slack :) Anything you can send me over would be much appreciated, I have no problems in giving you a pre-release copy so you can give some feedback too. Regards, Ben Merrills Griffin Internet T: 0870 8040862 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wayne Sheppard Sent: 14 October 2004 19:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (Another) Queue log analyser Very nice work Ben, thanks. Here are some additional thoughts - One segmentation that might be useful would be to add outbound calling activities as a either a separate column or even view. On agent stats, it would be useful to see login/logout stamps, login time, ready/not ready time (if this can be tracked, not sure). If you would like, I can send you some example reports that are used in a typical call center, contact me directly if you would find that helpful. Cheers, Wayne Ben Merrills wrote: 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limiting use of an account
Title: Limiting use of an account Dear Group, Could I limit the use of a VOIP account to restrict a user from making too many external calls? e.g. Could a user be given free access to the system would be allowed 20 mins of external calls a day? If this is possible could someone please supply me with an example of the required configuration? Warm Regards and Thanks Shad Mortazavi - Nexus Technical Manager n|m Nexus Management Inc Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limiting use of an account
Title: Limiting use of an account Thanks for the reply. I'm familiar with the use of contexts for user restrictions. I would really like to know if anyone had written an agi for this. Thanks Shad Mortazavi Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: RE: Creating conference calls from within Astman.
Title: RE: RE: Creating conference calls from within Astman. Sorry about my earlier e-mail. (:blush:), should have included the history. Can this be done from within Gastman? Warm Regards Shad -- Message: 2 Date: Wed, 22 Sep 2004 21:10:10 -0300 From: Nicol?s Gudi?o [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Creating conference calls from within Astman. To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hello, -Original Message- From: Shad Mortazavi Sent: Friday, September 17, 2004 1:03 PM To: [EMAIL PROTECTED] Subject: Creating conference calls from within Astman. Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a number of calls from within the conference room? I was incorporating this functionality into the Flash Operator Panel. In the panel I do it by using the Local channel. Suppose that you want to call the number 555-, and your context for dialing out is 'dialout'. You also have another context 'conferences' with extension number 1000 that fires up a meetme. The originate command for astman should look like: Action: Originate Channel: Local/[EMAIL PROTECTED] Exten: 1000 Context: conferences Priority: 1 It will dial the number 555 as if it were dialed from the dialout context and connect the call to Extension 1000 in 'conferences' context. This feature is already implemented and working in the next to be released version of the flash panel (but now it will only dial numbers predefined in the panel itself). You can get it from http://www.asternic.org Best regards, -- Nicolás Gudiño Buenos Aires - Argentina -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Creating conference calls from within Astman.
Title: RE: Creating conference calls from within Astman. Dear All, I sent this question a while back and was wondering if this was possible? Thanks Shad -Original Message- From: Shad Mortazavi Sent: Friday, September 17, 2004 1:03 PM To: [EMAIL PROTECTED] Subject: Creating conference calls from within Astman. Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a number of calls from within the conference room? Thanks for all your help. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RE: Creating conference calls from within Astman.
Title: Re: RE: Creating conference calls from within Astman. Thanks for this information. Can this be done from within Gastman as well? Warm Regards Shad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Creating conference calls from within Astman.
Title: Creating conference calls from within Astman. Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a number of calls from within the conference room? Thanks for all your help. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Logging into Multiple Call Queues on two * Servers and Voice Mail option.
Title: Logging into Multiple Call Queues on two * Servers and Voice Mail option. Dear All, I have two objectives that I need to meet; 1. I need to be able to log into two separate call queues on two different Asterisk servers, servicing two data centers. I seem to have problems configuring my SNOM phone to actively register with both servers. Has anyone got a working configuration for this? 2. I need to have an option for the user to press a button when in the call queue to go to voicemail. Has anyone got a working configuration for this? I appreciate all the help. Warm Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney
[Asterisk-Users] No Ringing.
Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney
[Asterisk-Users] Asterisk and SER Setup Questions.
Dear All, I have the following setup. Quad T1's-Asterisk (PBX)-(LAN-DMZ)-SER-(Firewall)-(Internet) | Local US Help Desk (Snom 200') This setup works well. I can pass calls from over the internet to the Asterisk PBX via SER using X-Ten Lit. I have a couple of questions; How do I tell Asterisk to forward all outbound URI calls to the SER proxy? This works for anyone on the ser itself, but what about someone on another system on the internet? Lets say I wanted to call someone at IPTEL.ORG? How do I tell Asterisk to forward calls that are not on it to ser? How do I append the caller ID so that my calls do not appear to come from Asterisk? Thanks and Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney
[Asterisk-Users] Losing my PRI Interface every 20-30 minutes???
Title: Losing my PRI Interface every 20-30 minutes??? Dear All, I'm having a problem with my Asterisk + E100P Installation in UK (BT PRI). The system functions as expected, and my dial plan works as expected. 30 minutes (or so) after starting the asterisk service I lose the PRI line, and only get this back after a service asterisk restart or reboot. During the failure there is no alarm on zttool, ztcfg show all 31 lines and there are no 'stuck' channels when I do a zap show channels. I'm using version 0.7.1 (need some agent features that I have patched into this copy), on Red Hat 9.0. I have had this combination system working in my data center in the USA since December with no problems. I'm not sharing an IRQ cat /proc/interrupts and there are no warnings in /var/log/messages. Here is a copy of my Zapata and Zaptel.conf files; The following options are switched on in my zaptel.conf; #loadzone = us #loadzone=it #loadzone=fr #loadzone=de loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no defaultzone=uk #Dornoch ISDN span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I have the following options in my Zapata.conf; language=en switchtype=euroisdn ; ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ; I have tried both national and unknown pridialplan=unknown ; overlapdial=yes signalling=pri_cpe ; usecallerid=yes ; threewaycalling=yes ; transfer=yes echocancel=yes ; echocancelwhenbridged=yes ; rxgain=0.0 txgain=0.0 ; group=1 ; callgroup=1 pickupgroup=1 immediate=no ; added this following a posting in news group busydetect=1 busycount=7 ; context=inbound channel = 1-15,17-31 I do see these messages in the CLI, but from what I hear they are common? CLI error messages; May 13 03:27:51 WARNING[1192437440]: chan_zap.c:5834 zt_pri_error: PRI: Read on 49 failed: Unknown error 500 May 13 03:27:53 WARNING[1192437440]: chan_zap.c:5834 zt_pri_error: PRI: Read on 49 failed: Unknown error 500 May 13 03:27:55 WARNING[1192437440]: chan_zap.c:5834 zt_pri_error: PRI: Read on 49 failed: Unknown error 500 I would appreciate any insights from anyone who may have had and resolved a similar problem. Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney
[Asterisk-Users] Problem with x-ten lite
Dear Group, At the moment I use SJPhone as my soft phone with Asterisk. I prefer the look and feel of the x-ten lite. However, when ever I use my x-ten lite I get a lot of breakup in my communication. E.g. I will play some hold music, and every 5-6 seconds I drop some packets. I dont have the same issue with SJPhone. Im sure this is a configuration issues, but I can work out where? Can someone point me in the right directions? Thanks and Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney
[Asterisk-Users] SoundPointR IP 300
Title: SoundPointR IP 300 Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney
[Asterisk-Users] Latency and 'Scratchy' Voice...
Title: Latency and 'Scratchy' Voice... Dear All, I have move from the USA to Sydney, Australia. I have gone from a data center environment at work and cable at home to a 513k/128k ADSL line. I'm experiencing two issues; 1) There is a latency of .5 - .8 seconds between me and the USA. 2) I have been in two calls where my voice has been describes as 'Scratchy'? I'm using a SIP Phone from SJ Phone, and a Plantronics USB Headset. In my Asterisk box I'm using the Quad T1 card. Any tips on how I could get around these two issues? I can understand the latency issue, what is contributing to the 'Scratchy' sound? I have not had this issue in the 4 months of running the product. Warm Regards and Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney
[Asterisk-Users] Asterisk Server Crashing with New Application
Title: Asterisk Server Crashing with New Application Dear All, I have been running a successful and very stable call center PBX based on 0.7.1 release. I need to be on this release because of a number of features that I have complied from 3rd party patches, for the call center. I will not be able to upgrade to release 1 until the patches catch up and I have done the required testing. The system was very stable until two days ago. The changes made were; 1) Installed a Second PBX in my second data center and I am running IAX2. 2) Installed the MySQL module. 3) Installed a copy of the php based CDR reporting. 4) Installed the Flash Operator Panel 5) Installed a modified version of Monastery to show me which agents were logged in and active I only stated having instability around the changes made in 4 and 5. I suspect the problem to be either caused by 4 or 5, in which case they will be very easy to rectify. I would however like to know if anyone else has had a) the same experience and b) has been able to isolate the issue. Warm Regards and Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney
[Asterisk-Users] RE: Asterisk Server Crashing with New Application
Chris, This does sound like my scenario. Do you remember how they achieved this? Now that I have removed these components Im stable again. Thanks for the feedback and help. Warm Regards Shad From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Saturday, April 10, 2004 4:46 AM To: [EMAIL PROTECTED] Cc: Shad Mortazavi Subject: Re: [Asterisk-Users] Asterisk Server Crashing with New Application Shad, I don't remember how far in the past, but a while back at least one person if not more reported instability in asterisk caused by more than one manager client connecting to the Asterisk server at the same time. Your monastery as well as the Flash Panel both access the manager application if my understanding of those applications is correct. The solution the person came up with was to put a single agent in front of the manager port to query the manager application on the Asterisk box and distribute the results to the client programs. You may have run into this same issue by running both flash operator and monastery. -Chris On 09:46 PM 4/8/2004, Shad Mortazavi wrote: snip 4) Installed the Flash Operator Panel 5) Installed a modified version of Monastery to show me which agents were logged in and active I only stated having instability around the changes made in 4 and 5.
[Asterisk-Users] Presence
Title: [Asterisk-Users] Presence I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney
[Asterisk-Users] IAX2 Problem and Question
Title: IAX2 Problem and Question Dear Asterisk Users. I have been setting up IAX between two servers, one in the USA and the other in UK so that I can pass help desk and general calls from one call center to another. I seem to be having an issue. When I set up IAX between my two servers I get into trouble when doing a reload on the CLI. Almost as if the system had gone into a loop reading the configuration. I also have a question, if I use the switch command i.e. switch = IAX2/brunswick:[EMAIL PROTECTED]/sip and switch = IAX2/dornoch:[EMAIL PROTECTED]/sip can I point to the same context? Or does the context on each pbx need to be unique. Thanks and Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney NSW 2089
[Asterisk-Users] Extension Questions
Dear all,
I have the following lines in my extentions.conf file;
;All US Calls
exten = _9001XX,1,Dial(IAX2/dornoch:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
;Dial 9 for outgoing numbers
exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
;include Brunswick
switch = IAX2/dornoch:[EMAIL PROTECTED]/sip
What Im trying to do is to send any calls starting
with 9001 out through my system in the USA and any number starting with a
9 through my local number.
However what ever the number I dial starting with a 9 goes
out of the local interface. If I comment out the exten
=_9.,1,Dial(Zap/g1/${EXTEN:1}) then it works.
How can I make the line starting exten = _9001 take precedence
over the line starting exten = _9001?
Kind Regards
Shad Mortazavi
-
US Technical Manager
Nexus Management
[Asterisk-Users] Troubles with the System Attendent Patch.
Title: Troubles with the System Attendent Patch. Dear all, I have spent some time tying to get the system attendant patch to work; http://bugs.digium.com/bug_view_page.php?bug_id=214 I get no errors patching the system and the function runs, but I keep getting the following error; queue: Nexus1, options: (null), url: (null), announce: (null), timeout: 0 -- Started music on hold, class 'default', on SIP/phone10-a3f0 -- Stopped music on hold on SIP/phone10-a3f0 Jan 23 15:19:04 WARNING[1226062640]: file.c:446 ast_openstream: File queue-youarenext (You are now first in line.) does not exist in any format Jan 23 15:19:04 WARNING[1226062640]: file.c:734 ast_streamfile: Unable to open queue-youarenext (You are now first in line.) (format ULAW): No such file or directory -- Told SIP/phone10-a3f0 in Nexus1 their queue position (which was 1) Jan 23 15:19:04 WARNING[1226062640]: file.c:446 ast_openstream: File queue-thankyou (Thank you for your patience.) does not exist in any format Jan 23 15:19:04 WARNING[1226062640]: file.c:734 ast_streamfile: Unable to open queue-thankyou (Thank you for your patience.) (format ULAW): No such file or directory I have put the files in exits in var/lib/asterisk/sounds. I'm running Redhat 9.0, Latest CVS, on a Dell 650. I have recompiled the Asterisk program several times and get to the same point. Any help or suggestions would be most welcome. Warm Regards Shad Mortazavi - US Technical Manager Nexus Management
[Asterisk-Users] System Attendent
Title: System Attendent Dear All, I have a number of call queues defined in Asterisk. I would like to program a system attendant that tells people; 1. Every 60 seconds 'Your call will be answered as soon as possible' 2. Tell the user how many calls are on the queue. I would then like them put back on hold music. Does someone have a configuration for this or something similar? Your help would be greatly appreciated. Kind Regards Shad Mortazavi US Technical Manager Nexus Management
[Asterisk-Users] Call Queue and Agent Statistics
Dear Group, I need to write a couple of reporting tools for my Call Center Asterisks implementation. I have multiple call queues with multiple agents that can sign in and based on gain access to multiple queues based on their assignments. I would like to write a script to collect call statistics for the agents the queues and the calls, and to put these into MySQL for reporting purposes. I'm thinking that each one of my customers would have their own table with the relevant information. Some of the statistics I'm looking for is; Which agents took the call. Average Call time Average Hold Time (How long was the call in the queue). In addition I'm looking at developing a couple of web pages; To show the agents that are logged into the system To show users that is registered. i.e. which interfaces are logged in and what is their status. I was wondering if anyone on this list was doing anything similar and would be able to share their ideas/code with me. I have written a number of large scale web based administration tools based on Perl and MySQL and would like to release this code to the Asterisk community once it is completed, to act as a call center tool. Warm Regards and Thanks --- Shad Mortazavi US Technical Manager Nexus Management
[Asterisk-Users] AgentCallbackLogin.
Dear Forum, I'm using the AgentCallbackLogin function to log my agents onto multiple call queues. exten = 3001,1, AgenCallbackLogin(1001,@sip). This works very well. I can not work out how to log them back out? On of the forum members was kind enough to point me into the directions of 'dial a null extension and press * to logout'. I don't seem to be able to translate this into Syntax. Can some one help? Warm Regards --- Shad Mortazavi US Technical Manager Nexus Management
[Asterisk-Users] Routing calls from a T1 based on DNSI.
Title: Routing calls from a T1 based on DNSI. Dear Group, I'm in the final phases of switching over from my existing PBX to an Asterisk based PBX. On my current PBX calls are routed on the existing PBX using a assigned DNSI number, and I'm looking at replicating this functionality. Does anyone have experience in routing calls from a T1 based on a DNSI number? If so would you mind; a) Confirming this functionality and b) giving me a sample of what this would look like in the configuration file? Warm Regards and Thanks --- Shad Mortazavi US Technical Manager Nexus Management
[Asterisk-Users] Agent setup
Dear Group, I have been successful in setting up the Agents, queues and getting agents to log in. Is there a way that I could configure the system so that the agent is called back. i.e. the agent logs into the system, a call is destined for them and their phone rings. If some one has this setup I would be very interested in hearing from them. Warm Regards and Thanks --- Shad Mortazavi US Technical Manager Nexus Management
