Re: [asterisk-users] Difference between Application Set and Function SET?
On Fri, Jun 16, 2017 at 1:43 PM, Jonathan Hwrote: > OK, thanks. That sort of makes sense. Is it case sensitive? > Is what case sensitive? Function names are case sensitive. Application names have historically been not case sensitive. > > Bonus quickie while I'm here (not worth own thread) - Asterisklint > complains that: > > H_PAT_NON_CANONICAL: pattern '_#' is not in the canonical form '#' > > for the line > > exten => _#,1,Goto(s,1) > > I'm sure I read somewhere it should be _#. > > Am I imagining it?! > You are declaring an extension line with a pattern but the pattern only has literal characters so it really isn't a pattern. It takes more CPU to match than the non-pattern form and is more likely an error. Richard [1] https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between Application Set and Function SET?
OK, thanks. That sort of makes sense. Is it case sensitive? Bonus quickie while I'm here (not worth own thread) - Asterisklint complains that: H_PAT_NON_CANONICAL: pattern '_#' is not in the canonical form '#' for the line exten => _#,1,Goto(s,1) I'm sure I read somewhere it should be _#. Am I imagining it?! On 16 June 2017 at 19:17, Richard Kennerwrote: >> It was only when I ran AsteriskLint over my dialplan that I noticed this: >> >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET >> >> Hmmm, they both seem to do the same thing. Or don't they? > > In some sense they do, but one's an application, meaning that it's > like a subprogram in a programming-language sense, and the other is a > function, which returns a value. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between Application Set and Function SET?
> It was only when I ran AsteriskLint over my dialplan that I noticed this: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET > > Hmmm, they both seem to do the same thing. Or don't they? In some sense they do, but one's an application, meaning that it's like a subprogram in a programming-language sense, and the other is a function, which returns a value. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between Application Set and Function SET?
On Fri, Jun 16, 2017 at 12:41 PM, Jonathan Hwrote: > It was only when I ran AsteriskLint over my dialplan that I noticed this: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET > > Hmmm, they both seem to do the same thing. Or don't they? > Yes they both do the same thing which is set a channel variable. However, when they can be invoked is different. The Set application can only be invoked in dialplan. The SET function can be invoked anywhere a function can be invoked and not just in dialplan. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
It looks like your database configuration is missing in Asterisk. It is making up information about the connection using defaault values as if it did not find any database configuration. Ron On 03/06/2013 10:49 AM, Olivier CALVANO wrote: Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com http://myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com http://myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com http://myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com http://myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd | forcename | forcegreetings | hidefromdir | stamp | +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ .. anyone know the problems ? thanks olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
The database schema (table) is different in Asterisk 11.4 ? because i have configured: cdr_mysql.conf extconfig.conf res_config_mysql.conf and on the mysql server, it's the old database of 1.6.x i see: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx (err 2003). Check debug for more info. can i put debug ? i don't know where thanks olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com It looks like your database configuration is missing in Asterisk. It is making up information about the connection using defaault values as if it did not find any database configuration. Ron On 03/06/2013 10:49 AM, Olivier CALVANO wrote: Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd | forcename | forcegreetings | hidefromdir | stamp | +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ .. anyone know the problems ? thanks olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
Strange too, in the logs: [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). [Jun 3 17:09:49] NOTICE[3464] config.c: Registered Config Engine mysql [Jun 3 17:09:49] ERROR[3464] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on (err 2003). Check debug for more info. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: Table Comptes_IAX not found in database. This table should exist if you're using realtime. Hi said No database host found but in the log i have Failed to connect database server SSI on with SSI and correct into my config file 2013/6/3 Olivier CALVANO o.calv...@gmail.com The database schema (table) is different in Asterisk 11.4 ? because i have configured: cdr_mysql.conf extconfig.conf res_config_mysql.conf and on the mysql server, it's the old database of 1.6.x i see: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx (err 2003). Check debug for more info. can i put debug ? i don't know where thanks olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com It looks like your database configuration is missing in Asterisk. It is making up information about the connection using defaault values as if it did not find any database configuration. Ron On 03/06/2013 10:49 AM, Olivier CALVANO wrote: Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd |
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
Fix this. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). Asterisk is telling you that you have not configured ANY database. It is not worrying about what tables are in it because you have not even defined the database itself. There is NO database at all so worrying about versions is not Asterisk's big problem.. The rest of the messages after that are a bit screwy because the routines producing the error are not aware that there is no database at all so they just complain about the piece that they know about. Ron On 03/06/2013 12:19 PM, Olivier CALVANO wrote: No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com mailto:o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com http://myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com http://myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com http://myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com http://myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd | forcename | forcegreetings | hidefromdir | stamp | +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ ..
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd | forcename | forcegreetings | hidefromdir | stamp | +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ .. anyone know the problems ? thanks olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn after in extconfig.conf: sipusers = mysql,general,Comptes_SIP sippeers = mysql,general,Comptes_SIP iaxusers = mysql,general,Comptes_IAX iaxpeers = mysql,general,Comptes_IAX extensions = mysql,general,Extensions meetme = mysql,general,MeetMe musiconhold = mysql,general,Musiconhold voicemail = mysql,general,VoiceMail and in cdr_mysql.conf [global] hostname=myhost.mydomain.net dbname=MyDB table=Cdr password=MyPassword user=MyUser port=3306 sock=/tmp/mysql.sock [aliases] start=calldate end=callend callerid=clid src=src dst=dst dcontext=dcontext channel=channel dstchannel=dstchannel lastapp=lastapp lastdata=lastdata duration=duration billsec=billsec disposition=disposition amaflags=amaflags accountcode=accountcode userfield=userfield uniqueid=uniqueid CodeTier=CodeTier you know what file I forgot to configure? Olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Fix this. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). Asterisk is telling you that you have not configured ANY database. It is not worrying about what tables are in it because you have not even defined the database itself. There is NO database at all so worrying about versions is not Asterisk's big problem.. The rest of the messages after that are a bit screwy because the routines producing the error are not aware that there is no database at all so they just complain about the piece that they know about. Ron On 03/06/2013 12:19 PM, Olivier CALVANO wrote: No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail;
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net http://myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn after in extconfig.conf: sipusers = mysql,general,Comptes_SIP sippeers = mysql,general,Comptes_SIP iaxusers = mysql,general,Comptes_IAX iaxpeers = mysql,general,Comptes_IAX extensions = mysql,general,Extensions meetme = mysql,general,MeetMe musiconhold = mysql,general,Musiconhold voicemail = mysql,general,VoiceMail and in cdr_mysql.conf [global] hostname=myhost.mydomain.net http://myhost.mydomain.net dbname=MyDB table=Cdr password=MyPassword user=MyUser port=3306 sock=/tmp/mysql.sock [aliases] start=calldate end=callend callerid=clid src=src dst=dst dcontext=dcontext channel=channel dstchannel=dstchannel lastapp=lastapp lastdata=lastdata duration=duration billsec=billsec disposition=disposition amaflags=amaflags accountcode=accountcode userfield=userfield uniqueid=uniqueid CodeTier=CodeTier you know what file I forgot to configure? Olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com Fix this. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). Asterisk is telling you that you have not configured ANY database. It is not worrying about what tables are in it because you have not even defined the database itself. There is NO database at all so worrying about versions is not Asterisk's big problem.. The rest of the messages after that are a bit screwy because the routines producing the error are not aware that there is no database at all so they just complain about the piece that they know about. Ron On 03/06/2013 12:19 PM, Olivier CALVANO wrote: No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com mailto:o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com http://myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com http://myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com http://myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com http://myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn after in extconfig.conf: sipusers = mysql,general,Comptes_SIP sippeers = mysql,general,Comptes_SIP iaxusers = mysql,general,Comptes_IAX iaxpeers = mysql,general,Comptes_IAX extensions = mysql,general,Extensions meetme = mysql,general,MeetMe musiconhold = mysql,general,Musiconhold voicemail = mysql,general,VoiceMail and in cdr_mysql.conf [global] hostname=myhost.mydomain.net dbname=MyDB table=Cdr password=MyPassword user=MyUser port=3306 sock=/tmp/mysql.sock [aliases] start=calldate end=callend callerid=clid src=src dst=dst dcontext=dcontext channel=channel dstchannel=dstchannel lastapp=lastapp lastdata=lastdata duration=duration billsec=billsec disposition=disposition amaflags=amaflags accountcode=accountcode userfield=userfield uniqueid=uniqueid CodeTier=CodeTier you know what file I forgot to configure? Olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Fix this. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). Asterisk is telling you that you have not configured ANY database. It is not worrying about what tables are in it because you have not even defined the database itself. There is NO database at all so worrying about versions is not Asterisk's big problem.. The rest of the messages after that are a bit screwy because the routines producing the error are not aware that there is no database at all so they just complain about the piece that they know about. Ron On 03/06/2013 12:19 PM, Olivier CALVANO wrote: No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
asterisk trying connect to mysql via socket remove that line from config files. 1 check if port 3306 is open in iptables on both servers. 2 check permissions on db for user Asterisk. On Mon, Jun 3, 2013 at 9:18 PM, Olivier CALVANO o.calv...@gmail.com wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn after in extconfig.conf: sipusers = mysql,general,Comptes_SIP sippeers = mysql,general,Comptes_SIP iaxusers = mysql,general,Comptes_IAX iaxpeers = mysql,general,Comptes_IAX extensions = mysql,general,Extensions meetme = mysql,general,MeetMe musiconhold = mysql,general,Musiconhold voicemail = mysql,general,VoiceMail and in cdr_mysql.conf [global] hostname=myhost.mydomain.net dbname=MyDB table=Cdr password=MyPassword user=MyUser port=3306 sock=/tmp/mysql.sock [aliases] start=calldate end=callend callerid=clid src=src dst=dst dcontext=dcontext channel=channel dstchannel=dstchannel lastapp=lastapp lastdata=lastdata duration=duration billsec=billsec disposition=disposition amaflags=amaflags accountcode=accountcode userfield=userfield uniqueid=uniqueid CodeTier=CodeTier you know what file I forgot to configure? Olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Fix this. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). Asterisk is telling you that you have not configured ANY database. It is not worrying about what tables are in it because you have not even defined the database itself. There is NO database at all so worrying about versions is not Asterisk's big problem.. The rest of the messages after that are a bit screwy because the routines producing the error are not aware that there is no database at all so they just complain about the piece that they know about. Ron On 03/06/2013 12:19 PM, Olivier CALVANO wrote: No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
Do you have this problem in your conf file? http://forums.digium.com/viewtopic.php?p=63736 The parser won't accept an ; (semicolon) for remarks! So he found at the first the old remarks and tried to access my database with the false data. Ron On 03/06/2013 3:18 PM, Olivier CALVANO wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com mailto:asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net http://myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn -- Ron Wheeler President Artifact Software Inc email:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
oh ron thanks for your help : We have deleted all commented line, only put the configuration and now that's work ! 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Do you have this problem in your conf file? http://forums.digium.com/viewtopic.php?p=63736 The parser won't accept an ; (semicolon) for remarks! So he found at the first the old remarks and tried to access my database with the false data. Ron On 03/06/2013 3:18 PM, Olivier CALVANO wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
grrr no in asterisk -d i have no error, but when i start normaly asterisk i have : [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net (err 2003). Check debug for more info. [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net (err 2003). Check debug for more info. what is the command in asterisk for i see the SQL query ? 2013/6/4 Olivier CALVANO o.calv...@gmail.com oh ron thanks for your help : We have deleted all commented line, only put the configuration and now that's work ! 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Do you have this problem in your conf file? http://forums.digium.com/viewtopic.php?p=63736 The parser won't accept an ; (semicolon) for remarks! So he found at the first the old remarks and tried to access my database with the false data. Ron On 03/06/2013 3:18 PM, Olivier CALVANO wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
Well, at least you are making progress. What is the error in the debug log? Ron On 03/06/2013 8:03 PM, Olivier CALVANO wrote: grrr no in asterisk -d i have no error, but when i start normaly asterisk i have : [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net http://xxx.xxx.net (err 2003). Check debug for more info. [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net http://xxx.xxx.net (err 2003). Check debug for more info. what is the command in asterisk for i see the SQL query ? 2013/6/4 Olivier CALVANO o.calv...@gmail.com mailto:o.calv...@gmail.com oh ron thanks for your help : We have deleted all commented line, only put the configuration and now that's work ! 2013/6/3 Ron Wheeler rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com Do you have this problem in your conf file? http://forums.digium.com/viewtopic.php?p=63736 The parser won't accept an ; (semicolon) for remarks! So he found at the first the old remarks and tried to access my database with the false data. Ron On 03/06/2013 3:18 PM, Olivier CALVANO wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com mailto:asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net http://myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn -- Ron Wheeler President Artifact Software Inc email:rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com skype: ronaldmwheeler phone:866-970-2435, ext 102 tel:866-970-2435%2C%20ext%20102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
What is the best commande for put the debug ? because with core set debug, i don't have a return. voip-2*CLI realtime mysql status Vop configured for m...@myhost.mydomain.net, port 3306 with username Asterisk. [Jun 4 06:48:25] ERROR[27879]: res_config_mysql.c:1577 mysql_reconnect: MySQL RealTime: Failed to connect database server Vop on vop.phibee-telecom.net (err 2003). Check debug for more info. He read correctly the config because it's the good DB, Server and username in /var/log/asterisk/message i have: [Jun 4 06:46:21] Asterisk 11.4.0 built by mockbuild @ buildvm-12.phx2.fedoraproject.org on a x86_64 running Linux on 2013-05-20 15:47:05 UTC [Jun 4 06:46:21] NOTICE[27825] loader.c: 1 modules will be loaded. [Jun 4 06:46:21] NOTICE[27825] config.c: Registered Config Engine mysql [Jun 4 06:46:21] NOTICE[27825] cdr.c: CDR simple logging enabled. [Jun 4 06:46:21] NOTICE[27825] loader.c: 192 modules will be loaded. [Jun 4 06:46:21] NOTICE[27825] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Jun 4 06:46:21] WARNING[27825] res_musiconhold.c: No music on hold classes configured, disabling music on hold. [Jun 4 06:46:21] ERROR[27825] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. [Jun 4 06:46:21] WARNING[27825] res_config_mysql.c: Table Comptes_IAX not found in database. This table should exist if you're using realtime. [Jun 4 06:46:21] ERROR[27825] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. [Jun 4 06:46:21] ERROR[27825] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. [Jun 4 06:46:21] WARNING[27825] res_config_mysql.c: Table Comptes_SIP not found in database. This table should exist if you're using realtime. [Jun 4 06:46:21] NOTICE[27825] confbridge/conf_config_parser.c: Adding default_user profile to app_confbridge [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: Starting AEL load process. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:22] ERROR[27825] cdr_mysql.c: Failed to connect to mysql database MyDB on MyHost.MyDomain.net http://myhost.mydomain.net/. [Jun 4 06:47:51] ERROR[27879] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. [Jun 4 06:48:25] ERROR[27879] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. Asterisk have the good information, but i don't understand why he can't connect to the DB, if i use: mysql -h MyHost.MyDomain.net http://myhost.mydomain.net/ -u Aserisk -p MyDB i have a full access to my MySQL server. may be missing in a Fedora package ? 2013/6/4 Ron Wheeler rwhee...@artifact-software.com Well, at least you are making progress. What is the error in the debug log? Ron On 03/06/2013 8:03 PM, Olivier CALVANO wrote: grrr no in asterisk -d i have no error, but when i start normaly asterisk i have : [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net (err 2003). Check debug for more info. [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net (err 2003). Check debug for more info. what is the command in asterisk for i see the SQL query ? 2013/6/4 Olivier CALVANO o.calv...@gmail.com oh ron thanks for your help : We have deleted all commented line, only put the configuration and now that's work ! 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Do you have this problem in your conf file? http://forums.digium.com/viewtopic.php?p=63736 The parser won't accept an ; (semicolon) for remarks! So he found at the first the old remarks and tried to access my database with the false data. Ron On 03/06/2013 3:18 PM, Olivier CALVANO wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure
Re: [asterisk-users] difference between playback and background?
Alternatively, if you don't have that extension defined anywhere, Asterisk will jump to the i extension, where you can then read the actual entered digits from the INVALID_EXTEN variable and jump back to the main part of the dialplan. Note that if they enter digits that *could* match a defined extension, Asterisk won't necessarily jump to the i extension until more digits are entered which may or may not cause no explicitly defined extensions to be matched. Cheers, Kingsley. On Mon, 2011-11-21 at 12:32 -0800, Steve Edwards wrote: On Mon, 21 Nov 2011, Danny Nicholas wrote: Option 2 Use WaitExten with Background [getnum] Exten = start,1,background(prompt) Exten = start,n,waitexten(2) Exten = ,1,noop(user pressed ) Exten = I,1,playback(invalid) For option 2 you have to define each valid 4 digit entry in the context. Or, (since the OP seems a bit newbish), read up on extension pattern matching. -- Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between playback and background?
It sounds like you may want to use the READ command instead. This lets you hard-set the number of digits to expect and then sets a variable which you can use later in the dialplan. Generally you use the background command to let them dial an extension or automated attendant option. Playback plays without the option to interrupt it. On Mon, Nov 21, 2011 at 10:50 AM, Edward de Jong edward.dej...@voicecarrier.com wrote: In the dial plan language of asterisk, what is the difference between prompting the user with a Playback() command vs. a Background() command? I want in a part of my dial plan to ask the user a prompt, and wait for 4 digits to be typed in. I don't want the user to have to end the string with a pound or something, just wait 2 seconds after they stop typing. ANd I do want the prompt to be interruptible if the user is fast and knows already what to do… I need to do some tests on the number they entered. If i use background(), and say the prompt, and then follow with a WAIT command, how do i reference the number they just typed in? does asterisk set the ${EXTEN} variable when the user types something? What I find maddening about the asterisk documentation is a lack of clarity on the sequence of things, and what variables get set when? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between playback and background?
First question - playback is not interruptable by DTMF, background is. You have two options here Option 1 Use Read [getnum] Exten = start,1,read(mydigit,prompt,4,skip,1,2) .. verification stuff Option 2 Use WaitExten with Background [getnum] Exten = start,1,background(prompt) Exten = start,n,waitexten(2) Exten = ,1,noop(user pressed ) Exten = I,1,playback(invalid) For option 2 you have to define each valid 4 digit entry in the context. Yes it can be maddening, but you get what you pay for. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Edward de Jong Sent: Monday, November 21, 2011 11:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] difference between playback and background? In the dial plan language of asterisk, what is the difference between prompting the user with a Playback() command vs. a Background() command? I want in a part of my dial plan to ask the user a prompt, and wait for 4 digits to be typed in. I don't want the user to have to end the string with a pound or something, just wait 2 seconds after they stop typing. ANd I do want the prompt to be interruptible if the user is fast and knows already what to do. I need to do some tests on the number they entered. If i use background(), and say the prompt, and then follow with a WAIT command, how do i reference the number they just typed in? does asterisk set the ${EXTEN} variable when the user types something? What I find maddening about the asterisk documentation is a lack of clarity on the sequence of things, and what variables get set when? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between playback and background?
On Mon, 21 Nov 2011, Danny Nicholas wrote: Option 2 Use WaitExten with Background [getnum] Exten = start,1,background(prompt) Exten = start,n,waitexten(2) Exten = ,1,noop(user pressed ) Exten = I,1,playback(invalid) For option 2 you have to define each valid 4 digit entry in the context. Or, (since the OP seems a bit newbish), read up on extension pattern matching. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between SIP peer and SIP user ?
Hi James, Thanks I give me the clear view and now I am able to describe that. again thank a lot 2011/5/23 James zhu zhulizh...@live.com hello: please refer this link: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com -- Date: Sat, 21 May 2011 17:49:37 +0530 From: virbh...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] difference between SIP peer and SIP user ? Hi list, I am confuse about these CLI commands *sip show users sip show peers* Can someone clear my doubt . what are the difference between them? - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between SIP peer and SIP user ?
Hi James, My question was related to CLI commands not sip.conf type value(friends,peer,user) main question was difference below commands *sip show peers* Vs *sip show user* On Mon, May 23, 2011 at 1:28 PM, virendra bhati virbh...@gmail.com wrote: Hi James, Thanks I give me the clear view and now I am able to describe that. again thank a lot 2011/5/23 James zhu zhulizh...@live.com hello: please refer this link: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com -- Date: Sat, 21 May 2011 17:49:37 +0530 From: virbh...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] difference between SIP peer and SIP user ? Hi list, I am confuse about these CLI commands *sip show users sip show peers* Can someone clear my doubt . what are the difference between them? - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between SIP peer and SIP user ?
hello: please refer this link: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Sat, 21 May 2011 17:49:37 +0530 From: virbh...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] difference between SIP peer and SIP user ? Hi list, I am confuse about these CLI commands sip show users sip show peers Can someone clear my doubt . what are the difference between them? - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
Thanks for sharing all of your thoughts and information. If anyone knows a good article about asterisk 1.8 then please let me know about it. I have read the presentation by Kevin Fleming but more information is always good. Cheers On Wed, Oct 6, 2010 at 10:28 AM, Miguel Molina mmol...@millenium.com.cowrote: I find 1.6.2.13 version is stable for trunk call routing, and it should be too for basic call center use. The asterisk team has made some architectural improvements (moving to astobj2 a lot of internal structures, and much more you may not see from a user perspective) but given the several environment and different use cases, fear to upgrade or proven 1.4 stability for the job, the people usually don't upgrade or make it slowly with a lot of previous tests before making the jump. If you use FAX, I recommend you 1.6.2 or later. The app_fax module is far better than the ast-agx-addons for 1.4. The good old (now unsupported) 1.2 works for many people, ask Steve. So it's up to you. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center El 06/10/10 11:04, Zeeshan Zakaria escribió: For a production environment, 1.4 is the most stable, and it has everything one needs to setup a telecom platform. As per my understanding 1.6 never got the same recognition for stability as 1.4, plus it doesn't have any significant advantages over 1.4. The newer version 1.8 series might be my next jump once it'll be out of beta, but at this time it should not be used in a production environment. Many of us still use 1.4 in production and if you are just starting, this'll be your best choice. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-06 11:54 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Be... *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham *Sent:* Wednesday, October 06, 2010 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Difference Is there any major architectural difference between 1.4 and 1.8? The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from , to |) and the AGI structure is enhanced. If you don’t use AGI’s, a qualified “not really”. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Qureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham Sent: Wednesday, October 06, 2010 7:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Difference Hi All, Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk versions. Thanks -- Best Regards Rizwan Qureshi In a nutshell, 1.4 is the oldest and most stable, 1.6 is the current and 1.8 is the beta version of Asterisk. This is a gross over-simplification, but if you know nothing, 1.4 is going to give you the fewest headaches and if you have to have the latest 1.6 or 1.8 is the way to go. The ChangeLogs on Asterisk.org will give you a detailed difference. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
Is there any major architectural difference between 1.4 and 1.8? On Wed, Oct 6, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham *Sent:* Wednesday, October 06, 2010 7:15 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Difference Hi All, Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk versions. Thanks -- Best Regards Rizwan Qureshi In a nutshell, 1.4 is the oldest and most stable, 1.6 is the current and 1.8 is the beta version of Asterisk. This is a gross over-simplification, but if you “know nothing”, 1.4 is going to give you the fewest headaches and if you “have to have the latest” 1.6 or 1.8 is the way to go. The ChangeLogs on Asterisk.org will give you a detailed difference. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Qureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham Sent: Wednesday, October 06, 2010 7:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Difference Hi All, Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk versions. Thanks -- Best Regards Rizwan Qureshi In a nutshell, 1.4 is the oldest and most stable, 1.6 is the current and 1.8 is the beta version of Asterisk. This is a gross over-simplification, but if you know nothing, 1.4 is going to give you the fewest headaches and if you have to have the latest 1.6 or 1.8 is the way to go. The ChangeLogs on Asterisk.org will give you a detailed difference. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham Sent: Wednesday, October 06, 2010 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Difference Is there any major architectural difference between 1.4 and 1.8? The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from , to |) and the AGI structure is enhanced. If you don't use AGI's, a qualified not really. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
On Wed, 6 Oct 2010, Rizwan Hisham wrote: Is there any major architectural difference between 1.4 and 1.8? Nope. The developer's just got tired of typing .4 Of course, the joke's on them -- 1.8 is only .4 better than 1.4. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
For a production environment, 1.4 is the most stable, and it has everything one needs to setup a telecom platform. As per my understanding 1.6 never got the same recognition for stability as 1.4, plus it doesn't have any significant advantages over 1.4. The newer version 1.8 series might be my next jump once it'll be out of beta, but at this time it should not be used in a production environment. Many of us still use 1.4 in production and if you are just starting, this'll be your best choice. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-06 11:54 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Be... *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham *Sent:* Wednesday, October 06, 2010 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Difference Is there any major architectural difference between 1.4 and 1.8? The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from , to |) and the AGI structure is enhanced. If you don’t use AGI’s, a qualified “not really”. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
Back in the days i heard that they have changed the architecture in 1.6 and its a lot better than 1.4 (6 times better call handling and robust architecture, someone told me). If they have decided to take the 1.6 architecture to the next level in the new 1.8 version then its a good thing. On Wed, Oct 6, 2010 at 9:58 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 6 Oct 2010, Rizwan Hisham wrote: Is there any major architectural difference between 1.4 and 1.8? Nope. The developer's just got tired of typing .4 Of course, the joke's on them -- 1.8 is only .4 better than 1.4. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Qureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
Here is a presentation from Kevin P. Fleming, Director of Software Technologies at Digium. Information might be old by now still gives a good overview of what is new in 1.6: http://www.asterisk-tag.org/2008/slides/Kevin-Fleming-Asterisk-Tag-2008.pdf Summary of his presentation is as follows: – Asterisk 1.6 contains much new functionality, although nothing revolutionary – Asterisk 1.6's core has been improved in many ways that will reduce the performance impact of new features being added and also the likelihood of difficult to find locking and data structure bugs – Future releases of Asterisk 1.6 (1.6.1, 1.6.2, etc.) will get new functionality as well, in a controlled fashion – Asterisk 1.6.0 is not recommended for production usage yet, but we would very much like users to try it, report problems and help test the product in more scenarios than the development can test themselves -- Zeeshan On Wed, Oct 6, 2010 at 12:12 PM, Rizwan Hisham rizwanhas...@gmail.comwrote: Back in the days i heard that they have changed the architecture in 1.6 and its a lot better than 1.4 (6 times better call handling and robust architecture, someone told me). If they have decided to take the 1.6 architecture to the next level in the new 1.8 version then its a good thing. On Wed, Oct 6, 2010 at 9:58 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 6 Oct 2010, Rizwan Hisham wrote: Is there any major architectural difference between 1.4 and 1.8? Nope. The developer's just got tired of typing .4 Of course, the joke's on them -- 1.8 is only .4 better than 1.4. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Qureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
I find 1.6.2.13 version is stable for trunk call routing, and it should be too for basic call center use. The asterisk team has made some architectural improvements (moving to astobj2 a lot of internal structures, and much more you may not see from a user perspective) but given the several environment and different use cases, fear to upgrade or proven 1.4 stability for the job, the people usually don't upgrade or make it slowly with a lot of previous tests before making the jump. If you use FAX, I recommend you 1.6.2 or later. The app_fax module is far better than the ast-agx-addons for 1.4. The good old (now unsupported) 1.2 works for many people, ask Steve. So it's up to you. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center El 06/10/10 11:04, Zeeshan Zakaria escribió: For a production environment, 1.4 is the most stable, and it has everything one needs to setup a telecom platform. As per my understanding 1.6 never got the same recognition for stability as 1.4, plus it doesn't have any significant advantages over 1.4. The newer version 1.8 series might be my next jump once it'll be out of beta, but at this time it should not be used in a production environment. Many of us still use 1.4 in production and if you are just starting, this'll be your best choice. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-10-06 11:54 AM, Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Be... *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham *Sent:* Wednesday, October 06, 2010 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Difference Is there any major architectural difference between 1.4 and 1.8? The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from , to |) and the AGI structure is enhanced. If you don’t use AGI’s, a qualified “not really”. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 'core show channels' and 'sip show channels' ??
On Sat, 7 Nov 2009, jonas kellens wrote: vps*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 0 active IAX channels vps*CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls vps*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message ip_peer(None) 58139462bde 00101/20006 0x0 (nothing)No Rx: REGISTER 1 active SIP channel You caught an endpoint registering - that is a channel while the conversation to register is occurring. vps*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message ip_peer 0a27b9c7d 37df36b96ba 04717/0 0x0 (nothing) No ip_peer (None) 24ddc4be2b2 00101/00119 0x0 (nothing)No Rx: REGISTER 2 active SIP channels One is a register again, not sure about the other. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
On Mon, Aug 3, 2009 at 4:29 PM, Michael Cunninghammichael.cunningham@gmail.com wrote: Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? This is asked so much that you can actually search the list for production use or stable. The answer isn't always answerable by anybody but you. The fact that you are trusted to make this decision for yourself is part of the asterisk philosophy. It depends on your time, aversion to risk, and willingness to determine whether a given version of asterisk meets your needs. You could argue that 'stable enough for production use' is an oxymoron. People either want the code to never change, so they never get features, or they want the latest and greatest, or they want the code to not break their outside enhancements, and of course it should be free, and vetted by somebody else first. Pick a few, and you'll be happy with the results. I'm personally running 1.6.2.0-beta4 in 'production' because certain features I want are more 'reliable' there. Other times I've pulled patches out of SVN, or reported my results on the issue tracker, again for the same reason. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
For changes between 1.4 and 1.6 you might find useful this one: http://svn.digium.com/svn/asterisk/tags/1.6.0/CHANGES For changes between 1.6 branches: http://svn.digium.com/svn/asterisk/tags/1.6.1.0/CHANGES http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta1/CHANGES Regards Jose 2009/8/4 Michael Cunningham michael.cunningham@gmail.com Thanks Leif, That cleared up the versioning.. Is there a list of new features in 1.6.x versus the 1.4.x version? On Mon, Aug 3, 2009 at 4:34 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: Michael Cunningham wrote: Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? Will Cepstral work with 1.6.x? This may be a useful article from asterisk.org for you to read: http://www.asterisk.org/node/48602 Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Cunningham TV Armor LLC, Owner 331 Fairfield Road, Bldg C-6 Freehold, NJ 07728 P: 1-800-890-0073 C: 732-618-6632 F: 732-414-2067 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
1.6.0 is stable 1.6.1 is stable 1.6.2 is release candidate See the files Changelog* and UPDATE* in this distributions for changes. regards klaus Michael Cunningham schrieb: Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? Will Cepstral work with 1.6.x? Thanks, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
Klaus Darilion wrote: 1.6.0 is stable 1.6.1 is stable 1.6.2 is release candidate See the files Changelog* and UPDATE* in this distributions for changes. Actually, almost... 1.6.0 is in release status 1.6.1 is in release status 1.6.2 is in beta status (soon release candidate) That's a more accurate way of describing them. Thanks! Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
Ok. What's the difference between stable and in release status? Does the hierarchy go beta, stable, release candidate, release status? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen Sent: Tuesday, August 04, 2009 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Difference between 1.4.x and 1.6.x? Klaus Darilion wrote: 1.6.0 is stable 1.6.1 is stable 1.6.2 is release candidate See the files Changelog* and UPDATE* in this distributions for changes. Actually, almost... 1.6.0 is in release status 1.6.1 is in release status 1.6.2 is in beta status (soon release candidate) That's a more accurate way of describing them. Thanks! Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
On Tuesday 04 August 2009 08:22:52 Danny Nicholas wrote: Ok. What's the difference between stable and in release status? Does the hierarchy go beta, stable, release candidate, release status? We don't actually use the word stable because it's a loaded word and leads to poor expectations. In Debian parlance, 1.2 would be considered stable (that is, does not change very often). The hierarchy goes from beta to release candidate to released. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
On Tue, Aug 04, 2009 at 08:45:13AM -0500, Tilghman Lesher wrote: On Tuesday 04 August 2009 08:22:52 Danny Nicholas wrote: Ok. What's the difference between stable and in release status? Does the hierarchy go beta, stable, release candidate, release status? We don't actually use the word stable because it's a loaded word and leads to poor expectations. In Debian parlance, 1.2 would be considered stable (that is, does not change very often). The hierarchy goes from beta to release candidate to released. But there's a missing characterization here: actively maintained. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
Michael Cunningham wrote: Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? Will Cepstral work with 1.6.x? This may be a useful article from asterisk.org for you to read: http://www.asterisk.org/node/48602 Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
Thanks Leif, That cleared up the versioning.. Is there a list of new features in 1.6.x versus the 1.4.x version? On Mon, Aug 3, 2009 at 4:34 PM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: Michael Cunningham wrote: Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? Will Cepstral work with 1.6.x? This may be a useful article from asterisk.org for you to read: http://www.asterisk.org/node/48602 Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Cunningham TV Armor LLC, Owner 331 Fairfield Road, Bldg C-6 Freehold, NJ 07728 P: 1-800-890-0073 C: 732-618-6632 F: 732-414-2067 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
Olivier wrote: Hello, Groups in asterisk are summarized here ( http://www.voip-info.org/wiki/view/Channels+and+Groups). Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789 (as I've been advised in another thread, to switch from one notation to the other and I can't see the reason behind that) ? Regards Assuming nothing has changed from Zaptel to DAHDI, the difference can found here: http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels Basically, the lowercase g chooses the lowest number available channel from the group where the uppercase G chooses the highest number available channel. This is used to reduce glare on analog or T1 (non-PRI) channels that are part of a hunt group. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
Olivier wrote: Hello, Groups in asterisk are summarized here (http://www.voip-info.org/wiki/view/Channels+and+Groups). Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789 (as I've been advised in another thread, to switch from one notation to the other and I can't see the reason behind that) ? Regards Using 'g' means that channels will be used starting with the lowest available number and then counting up. Starting with 'G' means to start at the highest channel number and count down. I believe that incoming calls will use the lowest numbered channels first, so if your outgoing calls start with the higher numbers it decreases the chance for glare on calls. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
Thanks for replying ! I've added a link here http://www.voip-info.org/wiki/view/Channels+and+Groups to http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels. 2008/12/1 Mark Michelson [EMAIL PROTECTED] Olivier wrote: Hello, Groups in asterisk are summarized here (http://www.voip-info.org/wiki/view/Channels+and+Groups). Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789 (as I've been advised in another thread, to switch from one notation to the other and I can't see the reason behind that) ? Regards Using 'g' means that channels will be used starting with the lowest available number and then counting up. Starting with 'G' means to start at the highest channel number and count down. I believe that incoming calls will use the lowest numbered channels first, so if your outgoing calls start with the higher numbers it decreases the chance for glare on calls. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between Asterisk and FreeSwitch
Please read: http://www.voip-info.org/wiki/view/FreeSwitch and http://www.voip-info.org/wiki/index.php?page=Asterisk Then if you have a specific question about one of them, come back here to ask about asterisk, and on the freeswitch mailinglist for more info on that technology. Or you could put up a comparision page on the voip-info wiki and ask folks to contribute there. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between Asterisk and FreeSwitch
On Tuesday 22 January 2008 11:50:25 love U.all wrote: what is the difference between FreeSwitch and Asterisk , The main difference in functionality is that FreeSwitch is a voip-switch only. It does not have any method to interface to the PSTN, other than through using another host which does have that connectivity, such as an Asterisk- based host. whitch one is more scalable and reliable? That is going to depend completely on what environment you're deploying it, what features you're using, etc. Keep in mind that Asterisk is going into its third major release cycle, while FreeSwitch is still undergoing public betas and has not yet had a single general release yet. Also, note that the installbase, developer base, and userbase are all much larger, by an exponential factor, for Asterisk than for FreeSwitch, and Asterisk has a company backing it which is willing to provide commercial support. FreeSwitch, as best as I can tell, has no such support structure. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between Asterisk and FreeSwitch
what is the difference between FreeSwitch and Asterisk , The main difference in functionality is that FreeSwitch is a voip-switch only. Technically, FreeSWITCH is a soft-switch, or a modular media switching library that can switch more than just voice. Also, technically, FS is a library, and there is a freeswitch application built on that library. The best analogy I can think of is the application curl, which is a command line app built on libcurl. Asterisk is a full-featured PBX that can do many of the things a true soft-switch can do. FreeSWITCH is (or will be, depending on your viewpoint) a full-featured soft-switch that can do many of the things that a PBX can do. It does not have any method to interface to the PSTN, other than through using another host which does have that connectivity, such as an Asterisk- based host. To be fair, this isn't quite accurate. FreeSWITCH can interface to PSTN via PRI or analog FXS/FXO using Digium, Sangoma, PIKA, etc. cards. (Any Zaptel-compatible cards should work. I've done PRI with a Tor2 clone.) Also, to be fair, the PSTN interface, like the rest of FS, is still young and therefore subject to the usual (and unusual) bugs that inhabit beta releases. Technically, the FreeSWITCH project is at RC1. The PSTN mod to FS is called OpenZAP and it is probably better described as beta. (Not an official statement, just my personal observation formed from my personal usage. I've got an Asterisk box sitting right next to a FS box and I've been playing with both of them and I can tell you that right now Asterisk is much more ready for PSTN usage.) whitch one is more scalable and reliable? That is going to depend completely on what environment you're deploying it, what features you're using, etc. Keep in mind that Asterisk is going into its third major release cycle, while FreeSwitch is still undergoing public betas and has not yet had a single general release yet. Also, note that the installbase, developer base, and userbase are all much larger, by an exponential factor, for Asterisk than for FreeSwitch, and Asterisk has a company backing it which is willing to provide commercial support. FreeSwitch, as best as I can tell, has no such support structure. These are all true. The bottom line is that FreeSWITCH is a young, but very cool, project headed by a small core development team. The lead developer is a huge Asterisk contributor - Anthony Minessale. (Check the karma page and I think you'll find he's way near the top...) The community is also small but growing quickly. There are a lot of people who use both FS and * because they have different target applications and different strengths and weaknesses. If you need a tried-and-true app that is well-supported and documented then Asterisk is an easy choice. If you are comfortable on the cutting edge or if you like the way FS is built or the way it approaches the handling of certain challenges then FS is something you should check out. This is one area where FOSS is so cool - you can totally check out both projects and give them a test drive without paying a penny in software costs. To the OP I recommend that you investigate both projects and see if one fits your needs better, which I believe is Tilghman's advice as well. -MC -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between TE121 and TE122
The TE121 is a PCI Express card (TE122 is standard PCI, 2.2 if I'm not mistaken). Gal Barak Tech support Atelis PLC 2008/1/16, Guilherme Loch Waltrick Góes [EMAIL PROTECTED]: What's the difference between the TE121 and TE122. I read the description on Digium's site and it isn't clear to me. Best regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between TE121 and TE122
Hello! Guilherme Loch Waltrick Góes wrote: What's the difference between the TE121 and TE122. I read the description on Digium's site and it isn't clear to me. Best regards, The only one difference is interface: one of them have PCI and other have PCI-Express. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between FXO interfaces !
The model AP200 that you are giving as example is 2 port only... and i'm not sure about the price... I know that codec conversion is one of the most cpu-intensive task that asterisk has to do, so, you can chose a Digium/Sangoma card with a powerful server doing the work or you can also use a VoIP gateway with a cheaper and less powerful asterisk box. It depens so much in your resources.. On 10/14/07, Mandeep Singh Bhabha [EMAIL PROTECTED] wrote: Hello everybody, Which one is a better choice 1. Gateway device with FXO - SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP. Office is having 12 phone lines. Thanks in Advance to all who shared his/her wisdom. -- With Regards, Mandeep Singh Bhabha email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between trunk and released versions
Yehavi, The release branches (1.2, 1.4) were at one time trunk. When it was decided to release 1.4, for example, it was branched off from trunk as the 1.4branch. New functionality continued to be added to trunk after that. Once the release branches are created, they are feature-frozen and only bug fixes are applied (this is the goal, though sometimes new functionality does sneak in when deemed necessary or desirable by the maintainers). So long story short, up until the release of 1.4, it was the same code as trunk. Since the 1.4 branch was created, no new functionality has been added to that branch, but 1.4.x releases continue to be made as bugs are discovered and fixed. Does this clear things up? Sean On Thu, 11 Oct 2007 16:34 +0200, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, Up to a while ago I thought that the released versions are checkpoints of the trunk versions; however, now I understand they are not, as I see differences between the two trains. So, what is the relation between them? Examples for differences: - When the language is different than Engligh the trunk version is reading numbers from /var/lib/asterisk/sounds/Lang-Name/digits while the release version is using /var/lib/asterisk/sounds/digits/Lang-Name - MAILBOX_EXISTS function is replaced with MailboxExists application. - External IVR has no way to exit from the program under the release version... The documentation is correct with the trunk version. Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Any fool can write code that a computer can understand. Good programmers write code that humans can understand. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between trunk and released versions
Hello Sean, Does this clear things up? Yes! Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference in show channels
'show channels' shows only running calls while 'sip show channels' shows all running sip sessions including phones trying to register . On 09/09/2007, ram [EMAIL PROTECTED] wrote: Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference in show channels
On 9/9/07, Jaswinder Singh [EMAIL PROTECTED] wrote: 'show channels' shows only running calls while 'sip show channels' shows all running sip sessions including phones trying to register . thanks but after my 30 channels of show channels i see lot of vice break and choppy voice doing passthrough codecs Xeon 2.0GHZ with 2 GG Ram centos 4.4 1.2.17 any suggestions ram On 09/09/2007, ram [EMAIL PROTECTED] wrote: Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between WaitExten and TIMEOUT (response)
Michiel van Baak wrote: On 05:27, Fri 03 Aug 07, bilal ghayyad wrote: Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? WaitExten is waiting for you to type an extension. TIMEOUT is to set the default timeout for promtps in IVR and stuff but is not actually waiting for you to provide an extension More specifically, timeout is the time between dialing digits when using WaitExten or background for asterisk to decide you are done dialing an option or extension and place the call. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between WaitExten and TIMEOUT (response)
The difference is in the scope of the command. Think of it this way: WaitExten gives the user more time to enter digits before the dialplan moves on to the next instruction in the dial plan. Timeout is the max number of seconds to wait at any point in the current context before deciding the user either got confused, doesn't know what their doing, or fell asleep. Clear as mud? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Friday, August 03, 2007 8:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Difference between WaitExten and TIMEOUT (response) Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between WaitExten and TIMEOUT (response)
Dear Michael; I understood it in that way (please advise me if I am correct): WaitExten is for the time to complete entering the digits, while timeout is specified wether user responded by dialing any thing or not. Please advise. regards The difference is in the scope of the command. Think of it this way: WaitExten gives the user more time to enter digits before the dialplan moves on to the next instruction in the dial plan. Timeout is the max number of seconds to wait at any point in the current context before deciding the user either got confused, doesn't know what their doing, or fell asleep. Clear as mud? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Friday, August 03, 2007 8:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Difference between WaitExten and TIMEOUT (response) Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between WaitExten and TIMEOUT (response)
On 05:27, Fri 03 Aug 07, bilal ghayyad wrote: Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? WaitExten is waiting for you to type an extension. TIMEOUT is to set the default timeout for promtps in IVR and stuff but is not actually waiting for you to provide an extension -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between WaitExten and TIMEOUT (response)
On 8/3/07, bilal ghayyad wrote: Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? core show function TIMEOUT for different timeout parameters, I haven't used WaitExten -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between SCCP and Cisco Call Manager traffic?
Call setup/teardown is handled with the SIP protocol while the actual call audio is handled with RTP I think. Check the config of your NAT devices relative to RTP. scd On 4/16/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'm wondering about the difference between Cisco Call Manager and SCCP(2) network traffic. I'm working on getting a Cisco 7960 phone to speak through a NAT to an asterisk box, without having to do a bunch of port forwarding on the NAT device. Without the nat, everything works fine. If the phone is behind a cisco pix that is doing the natting, it works fine (fixup protocol). If the phone is behind a more generic nat device, such as a linux box running ipfilter. Then it can dial out, but there is no audio. The interesting part is that this same phone, behind the same NAT works just fine if it is talking to a Cisco Call Manager box instead of an asterisk server. So, I'm wondering what the difference in the protocols is (I no longer have access to the call manager box, so I can't look @ the traffic). In a perfect world, I'd like to have the phone pretty much just work wherever it's plugged in as long as it can see the asterisk server. Any ideas ? Thanks Shawn ___ --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 -- Steve Dickey Who is John Galt? ___ --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: [Asterisk-Users] difference betwen a TE411P and TE410P
Rodney G. McDuff wrote: Is the TE411P just a TE410P with hardware echo cancellation? Yes. Same for a TE405P and TE406P. ___ --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: [Asterisk-Users] difference between records in CDR and realduration of call
On 3/10/06, AR Tarzi [EMAIL PROTECTED] wrote: That's because the duration is counted from the time of dialling. billsec is what you want if it's to calculate the duration the call was active. To change what shows you need to change call-log.php in /var/www/html/admin/cdr/ Instead of duration extract billsec - you can still label it duration (as in the title of the column). It might be closer to what you wish. - Original Message - From: nik600 [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 10, 2006 17:04 Subject: [Asterisk-Users] difference between records in CDR and realduration of call hi i've made some test calls, i've notice that a call of the duration of 1:29 minutes is recorded in the cdr database as 1:45 minutes, is it normal? i think that 15 seconds are too many... how can i correct this? thanks ___ --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 ___ --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 ok thanks, i've tried to edit the file but the field duration is set many times...exactly, where have i to cheange the value? thanks ___ --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: [Asterisk-Users] difference between records in CDR and real duration of call
What about ResetCDR() just before Dial()? nik600 wrote: hi i've made some test calls, i've notice that a call of the duration of 1:29 minutes is recorded in the cdr database as 1:45 minutes, is it normal? i think that 15 seconds are too many... how can i correct this? thanks ___ --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 ___ --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: [Asterisk-Users] difference between records in CDR and real duration of call
On Fri, 10 Mar 2006 15:04:18 +0100 nik600 [EMAIL PROTECTED] wrote: hi i've made some test calls, i've notice that a call of the duration of 1:29 minutes is recorded in the cdr database as 1:45 minutes, is it normal? i think that 15 seconds are too many... how can i correct this? thanks Maybe you answer the call in the dialplan before you pickup? Please show us the dialplan you used... and do you mean duration or billsec? regards christian ___ --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: [Asterisk-Users] difference between records in CDR and real duration of call
Or maybe, you should try the C flag in your Dial(). []'s MM nik600 wrote: hi i've made some test calls, i've notice that a call of the duration of 1:29 minutes is recorded in the cdr database as 1:45 minutes, is it normal? i think that 15 seconds are too many... how can i correct this? thanks ___ --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 ___ --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: [Asterisk-Users] difference between records in CDR and real duration of call
ok, thanks for your reply, tomorrow i'll test and let you know bye ___ --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: [Asterisk-Users] difference between records in CDR and realduration of call
That's because the duration is counted from the time of dialling. billsec is what you want if it's to calculate the duration the call was active. To change what shows you need to change call-log.php in /var/www/html/admin/cdr/ Instead of duration extract billsec - you can still label it duration (as in the title of the column). It might be closer to what you wish. - Original Message - From: nik600 [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 10, 2006 17:04 Subject: [Asterisk-Users] difference between records in CDR and realduration of call hi i've made some test calls, i've notice that a call of the duration of 1:29 minutes is recorded in the cdr database as 1:45 minutes, is it normal? i think that 15 seconds are too many... how can i correct this? thanks ___ --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 ___ --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: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
On Mon, Jul 18, 2005 at 12:03:42AM -0500, Kristian Kielhofner wrote: trixter http://www.0xdecafbad.com wrote: On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote: OT: Not a Knoppix, actually. You can't do anything useful with it without a HD install. A while ago I needed badly to test a certain system with Asterisk without installing it and was amazed to see the little existing support LiveCDs had of Asterisk. I ended up using AsteriskLive 0.1.6, even though it was rather old. If you wanted a knoppix install there is knopsterix, which can save your config without overwriting your whole drive without prompting you to save a partition or two. However they were not avilable for download, so I never bothered. I should be releasing a much improved Live version of AstLinux within a week or so. A test version was announced on my mailing list a while ago, with pretty good results so far. It will be AstLinux 0.2.8, and available as an ISO (as well as the Windows install package, disk images, etc.) I'll let everyone here know when it's released. This was actually the ifrst one I tried. I tried astlinux 0.2.6 as well, however as a live CD it wasn't useful, because I could not write any modified configuration on the live /etc . Thus I could not run asterisk with the modified configuration without a proper installation. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
Tzafrir Cohen wrote: I should be releasing a much improved Live version of AstLinux within a week or so. A test version was announced on my mailing list a while ago, with pretty good results so far. It will be AstLinux 0.2.8, and available as an ISO (as well as the Windows install package, disk images, etc.) I'll let everyone here know when it's released. This was actually the ifrst one I tried. I tried astlinux 0.2.6 as well, however as a live CD it wasn't useful, because I could not write any modified configuration on the live /etc . Thus I could not run asterisk with the modified configuration without a proper installation. Not quite... By default, AstLinux (and the live cd) will copy a default Asterisk configuration to a tmpfs filesystem where you can edit it, you just can't save it anywhere (persistent across reboot) unless you use a key disk. /etc/asterisk is in fact just a link to /tmp/etc/asterisk. Did you actually try to write a file in /etc/asterisk, or did you see that / was read-only and give up? Did you read the user guide? However, the main changes with the 0.2.7 and later ISO image are that the CD device is auto detected and the entire contents of the CD are copied to RAM so that the configuration files can be edited and the CD can be removed, that should make it more of a familiar live cd environment. -- Kristian Kielhofner ___ 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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
On Mon, Jul 18, 2005 at 02:40:40AM -0500, Kristian Kielhofner wrote: I tried astlinux 0.2.6 as well, however as a live CD it wasn't useful, because I could not write any modified configuration on the live /etc . Thus I could not run asterisk with the modified configuration without a proper installation. Not quite... By default, AstLinux (and the live cd) will copy a default Asterisk configuration to a tmpfs filesystem where you can edit it, you just can't save it anywhere (persistent across reboot) unless you use a key disk. /etc/asterisk is in fact just a link to /tmp/etc/asterisk. Did you actually try to write a file in /etc/asterisk, or did you see that / was read-only and give up? Did you read the user guide? Hmm, I remember I did. But it was a while ago... However, the main changes with the 0.2.7 and later ISO image are that the CD device is auto detected and the entire contents of the CD are copied to RAM so that the configuration files can be edited and the CD can be removed, that should make it more of a familiar live cd environment. That's even better. I'll try it next time. Thanks -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
My 2c worth... For the beginner, AAH is great. The PC that you install on will be totally reformatted / fdisk-ed (assuming single drive - etc). With AAH 1.3 - the installation goes to sleep and sort of finishes when its Syncing with a Time Server. A reboot at this point seems to do no harm. As Asterisk is configured via AMP - you are limited in functionality as to what AMP can do for you - but one can edit the config files directly as well for custom configs. (I needed to program an incoming (Fax) zap line to go to one particular extension) As it starts - there are a number of dialplan features which are quite cool, eg Time, Weather, Wakeup-Call, You extension is.., VoiceMail, IVR, Do-Not-Disturb, FAX handling. Sure - these are all things Asterisk can do, but with the default asterisk download, you start with a pretty clean slate... My current AAH limitations include:- a) In IVR, no ability to program or hold for an operator timeout for the DTMF challenged. b) Support for junghanns cards (or HFC cards) c) Multi-Company support - Default is one primary IVR Just did an install with many extensions 16 lines (4 x TDM400P) - 2 to Fixed line Cells, 14 to Telco (no services except DTMF dialing - its in East Africa). -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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
RE: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
my $0.02... I have implemented [EMAIL PROTECTED] 0.9 ina Real Estate/Legal office with 6 phones and 3 incoming lines using TDM04B card and Polycom/Sipura phones. I am about to implement 3 more systems ([EMAIL PROTECTED] 1.3) for a construction company.One will be 6 phones and 4 PSTN lines using SPA-3000 ATA's. The second will be 2x TDM04B connecting to 6 lines to start with10 Polycom IP-501 phones. The third system has not yet been designed. but will be 4 lines and 6 - 7 phones. The Real Estate system is running [EMAIL PROTECTED] 0.9 and has been running for 59 days since the last reboot, which I did when I added the Polycom phones. This is a fairly light-use system averaging about 1700 minutes/mo in usage. It is setup to have an Auto-Attendant when no one is present and to have the phones answered live when the receptionist is at her desk. I setup a "* + number" code for the receptionist to dial to toggle the Auto-attendant. As of yet I have not run into any problems running the one system in production, and we are confident in the other 3 systems that are about to be implemented. The systems that will be implemented soon will have pretty high call volume, as these are for a construction job trailers on pretty large scale projects. I started off by exploring Asterisk and actually built a couple of test systems without using [EMAIL PROTECTED]. I was able to figure out the configurations and actually had no problems setting up thetest systems (for me it was pretty simple even though I had never touched linux before). I created a set of config files that I was going to copy from server to server to aid the implementation, and I was ready to build the server for the Real Estate office... Then, I ran across [EMAIL PROTECTED] and after playing with it a little bit determined that it provided a quicker setup process and provided all the features that we would need. Clearly, there are some things that are easier to do without [EMAIL PROTECTED] due to the limitations provided by AMP... but I have found that for these small office implementations, this has not been an issue. Hope this helps. Jeff Busch [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael FelderSent: Sunday, July 17, 2005 8:17 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ 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
Re: [Asterisk-Users] Difference between Asterisk and Asterisk@home
That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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
RE: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
I guess I was wondering if it was crippled in some way. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Monday, 18 July 2005 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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 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
RE: [Asterisk-Users] Difference between Asterisk and Asterisk@home
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Monday, 18 July 2005 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. It should be mentioned that the [EMAIL PROTECTED] CD-ROM will silently (...or with very little fanfare) blow away any partitions you may have on your PC and install Linux and [EMAIL PROTECTED] So Be very carefull is booting on the [EMAIL PROTECTED] CD-ROM on a system that you don't want to sacrifice. That said, I have several [EMAIL PROTECTED] installs and they work fine for my purposes. I have dual TDM400P's with 8 incoming PSNT (TDM) lines from the Telco, and a combination of SIP and IAX2 hard and softphones. On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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 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
RE: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
Not in any way that you would be worried about for at least a while. It does restrict what you can customize but by the time you get to that stage (I still haven't - you can install your own version) Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Felder Sent: Sunday, 17 July 2005 11:33 PM To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] I guess I was wondering if it was crippled in some way. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Monday, 18 July 2005 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED] That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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 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 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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
I think that AAH would suit you fine for a small business application. It is not crippled in any way. Far from it in fact. It has more features installed as default than the latest CVS version. It is however based on the latest official release of the Asterisk code (whatever that is at the time you download the disk) and so may not have some of the latest bug fixes etc. Go for it! I've install a few for my customers and they love it. Mark Michael Felder wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
Michael Felder wrote: Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike [EMAIL PROTECTED] is a Linux distribution that makes it easy to install Asterisk and a few other Asterisk related applications (like AMP). Asterisk is the main application that makes [EMAIL PROTECTED] different from the CentOS distro (which is what it is based off of). Your question is a little like what is the difference between Apache and Fedora (Fedora usually includes Apache on server installs). That being said, [EMAIL PROTECTED] is crippled, but only because by default is uses AMP, which somewhat limits what you can change via it's interface. -- Kristian Kielhofner ___ 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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
Mark Phillips wrote: I think that AAH would suit you fine for a small business application. It is not crippled in any way. Far from it in fact. It has more features installed as default than the latest CVS version. Not to sound rude, but that doesn't make any sense. Or I'm reading it wrong :). [EMAIL PROTECTED] is a distro that by default includes a release version of Asterisk STABLE. First of all, you can't even directly compare the two (Asterisk vs. [EMAIL PROTECTED]). Would you compare an engine by itself to an entire car? No, you wouldn't. Don't do it here, either. Secondly, unless they are doing some mad branching and patching, there is NO WAY that the version of stable that they include has more Asterisk features than CVS HEAD. They include some applications that help enable some Asterisk features (mysql, mpg123, etc), but there is no reason that you can't get all of that with another distro (albeit not as easily) and Asterisk stable, or even more Asterisk features with all of those applications and CVS HEAD. It is however based on the latest official release of the Asterisk code (whatever that is at the time you download the disk) and so may not have some of the latest bug fixes etc. Go for it! I've install a few for my customers and they love it. Great! Use what works! Mark -- Kristian Kielhofner ___ 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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote: That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. OT: Not a Knoppix, actually. You can't do anything useful with it without a HD install. A while ago I needed badly to test a certain system with Asterisk without installing it and was amazed to see the little existing support LiveCDs had of Asterisk. I ended up using AsteriskLive 0.1.6, even though it was rather old. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote: On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote: That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. OT: Not a Knoppix, actually. You can't do anything useful with it without a HD install. A while ago I needed badly to test a certain system with Asterisk without installing it and was amazed to see the little existing support LiveCDs had of Asterisk. I ended up using AsteriskLive 0.1.6, even though it was rather old. If you wanted a knoppix install there is knopsterix, which can save your config without overwriting your whole drive without prompting you to save a partition or two. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
trixter http://www.0xdecafbad.com wrote: On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote: On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote: That question could start a battle. [EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the Knoppix of the Linux distro world. It could work but that is for you to decide. OT: Not a Knoppix, actually. You can't do anything useful with it without a HD install. A while ago I needed badly to test a certain system with Asterisk without installing it and was amazed to see the little existing support LiveCDs had of Asterisk. I ended up using AsteriskLive 0.1.6, even though it was rather old. If you wanted a knoppix install there is knopsterix, which can save your config without overwriting your whole drive without prompting you to save a partition or two. I should be releasing a much improved Live version of AstLinux within a week or so. A test version was announced on my mailing list a while ago, with pretty good results so far. It will be AstLinux 0.2.8, and available as an ISO (as well as the Windows install package, disk images, etc.) I'll let everyone here know when it's released. -- Kristian Kielhofner ___ 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
Re: [Asterisk-Users] Difference between Asterisk and Asterisk@home?
Kib Eki wrote: Hi, can one summarize the main differences between Asterisk and [EMAIL PROTECTED] or point me to a location where i can find such a list? See [EMAIL PROTECTED] site - http://asteriskathome.sf.net - and asterisk site - www.asterisk.org. Basically, asterisk is a program, and [EMAIL PROTECTED] is a distribution with running (and partially configured) asterisk, AMP, etc. and other additional stuff. Of course you have to configure your asterisk hardware yourself. It's like a question: what's the difference between KDE and Debian.. :) Tomek -- Znajdz swoja milosc na wiosne... http://link.interia.pl/f187a ___ 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
Re: [Asterisk-Users] Difference Between NAT=yes and QUALIFY=yes and STUN...
Matt wrote: I have a STUN server running on my Asterisk box which seems to work for most of my SIP clients.. but some of them seem to require NAT=yes turned on. If I go further and turn QUALIFY=yes to on, is there a reason I need to keep running a STUN server? If so, what's the difference? I never understood why Asterisk users seem to have such a fetish for STUN and SER. Most people don't need them. If you have many phones behind NAT and you want the phones to call each other and you want to enable reinvites then, yes, you need SER or STUN or something like that. Asterisk seems to be commonly used in three ways: 1) Home Phone System 2) Business Phone System 3) Internet Telephony Service Provider Generally none of these types of use has a large percentage of phones behind NAT and calling each other. Companies like FWD, etc DO need this since most of their users are calling each other. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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
Re: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
Am Samstag 05 März 2005 07:58 schrieb Remco Barende: Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no voip) while Snom has several other models that are IP. Who's cloning who? I don't want to end up with phones for which firmware support or update will disappear soon while the 'orginal' will continue to be supported? Can't say anything about support, but my personal research told me that they are the same - no even more they are identical. Jens ___ 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
Re: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
On 5 Mar 2005, at 06:58, Remco Barende wrote: Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no voip) while Snom has several other models that are IP. Who's cloning who? I don't want to end up with phones for which firmware support or update will disappear soon while the 'orginal' will continue to be supported? The way I heard it was that Snom had some trouble with the mechanical design of their earlier phones so bought the case design in from an existing ISDN phone maker. I guess that must be Elmeg. I guess the outer look tells you nothing about the hardware let alone the firmware. Tim. ___ 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
Re: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
Hello Remco, On Sat, 5 Mar 2005, Remco Barende wrote: Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no voip) while Snom has several other models that are IP. Who's cloning who? I don't want to end up with phones for which firmware support or update will disappear soon while the 'orginal' will continue to be supported? Elmeg has been for a long time a manufacturer of ISDN Phones and small to medium PBXes in germany. Snom uses the chassis of the elmeg phones and puts their own electronics in them. So it seems very likely that the elmeg IP-Phones are in fact Snom phones. I do not wether the firmware can be changed across elmeg and snom, but if there are no artificial barriers in place that prevent this this could be possible. Torsten Thx! Remco ___ 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 -- Media Online Internet Services Marketing GmbH Torsten Krueger [EMAIL PROTECTED] fon: 49-231-5575100fax: 49-231-55751098 Kurze Str. 10 D-44137 Dortmund ___ 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
RE: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
There is a partnership between Elmeg and snom. We are using their plastic (in the snom 190/200/220), they are using our hard- and software (in the Elmeg 290). Elmeg have a long experience in making phones and we have experience in making hard- and software for VoIP (as long as it can be in the SIP-based industry). A good partnership! We call it snom inside... CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Torsten Krueger Sent: Saturday, March 05, 2005 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between Snom 190 Elmeg 290? Hello Remco, On Sat, 5 Mar 2005, Remco Barende wrote: Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no voip) while Snom has several other models that are IP. Who's cloning who? I don't want to end up with phones for which firmware support or update will disappear soon while the 'orginal' will continue to be supported? Elmeg has been for a long time a manufacturer of ISDN Phones and small to medium PBXes in germany. Snom uses the chassis of the elmeg phones and puts their own electronics in them. So it seems very likely that the elmeg IP-Phones are in fact Snom phones. I do not wether the firmware can be changed across elmeg and snom, but if there are no artificial barriers in place that prevent this this could be possible. Torsten Thx! Remco ___ 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 -- Media Online Internet Services Marketing GmbH Torsten Krueger [EMAIL PROTECTED] fon: 49-231-5575100fax: 49-231-55751098 Kurze Str. 10 D-44137 Dortmund ___ 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 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
Re: [Asterisk-Users] Difference between E1 and PRI
PRI comes in 2versions E1 European and T1 US E1 30 channels T1 23 channels On Wed, 2005-02-23 at 14:15, Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ 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 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
Re: [Asterisk-Users] Difference between E1 and PRI
Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? Think of E-1 as Ethernet (transport) and PRI as IP (protocol). You could also think of E-1 as IP and PRI as TCP. You can also think of E-1 as TCP/IP and PRI as FTP. ___ 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
RE: [Asterisk-Users] Difference between E1 and PRI
E1 is a European T1. T1/E1 is the transport. PRI is the protocol. PRI on an T1 id 23B+D, PRI on an E1 is 30B+D. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, February 23, 2005 7:50 AM To: Eric Bishop; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between E1 and PRI Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? Think of E-1 as Ethernet (transport) and PRI as IP (protocol). You could also think of E-1 as IP and PRI as TCP. You can also think of E-1 as TCP/IP and PRI as FTP. ___ 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 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
Re: [Asterisk-Users] Difference between E1 and PRI
On Wed, 23 Feb 2005, Eric Bishop wrote: I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? E1 is a serial line capable of 2048 Mbit. After channelization you have 31 usable 64 kbit channels. One channel (number 16) is used for signalling (even when using CAS) and the remaining 30 channels are available for voice. When you run ISDN PRI over the E1 the ISDN signalling is placed in the signalling channel of the E1. Peter ___ 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
Re: [Asterisk-Users] Difference between E1 and PRI
Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ 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 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
Re: [Asterisk-Users] Difference between E1 and PRI
Alistair- Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to take a call back and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR. Regards, Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Alistair Cunningham wrote: Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ 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 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 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
RE: [Asterisk-Users] Difference between E1 and PRI
Scott, Do a search on Tromboning I have no idea if asterisk is capable of doing this but I remember this was a feature introduce into Fujitsu Qsig stack in or about 94-95 which solved a heap of customer problems at the time so I remember it was a big deal. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Wednesday, February 23, 2005 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between E1 and PRI Alistair- Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to take a call back and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR. Regards, Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Alistair Cunningham wrote: Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ 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 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 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 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
Re: [Asterisk-Users] Difference between E1 and PRI
Scott, Yes, and this is one of the principal reasons people choose Q.SIG. I've worked on quite a few large voicemail servers, and these tend to do a lot of transfers for follow-me and operator features. Q.SIG support can significantly reduce the number of telephony channels needed, as not only are there zero channels in use rather than two during the transfer, but transferred calls last significantly longer on average than calls to leave or retrieve messages. You do need to check that the remote end supports this; some older PBXs only support parts of the Q.SIG standard. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Scott Stingel wrote: Alistair- Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to take a call back and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR. Regards, Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Alistair Cunningham wrote: Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ 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 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 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 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
Re: [Asterisk-Users] Difference between E1 and PRI
On Wed, 23 Feb 2005, Scott Stingel wrote: Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to take a call back and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR. That problem goes by many names: tromboning, hairpinning, etc. There are several signalling methods, depending on what protocol is spoken by the remote switch: * q.sig has (can have) support for this * Explicit Call Transfer (ECT), used on EuroISDN I think) * 2B Channel Transfer, on 5ESS switches I don't think these are signalled the same way. Peter ___ 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