[Asterisk-Users] incomplete address
Hello I have configured asterisk with getting user authentication from MySQL. Now the problem is when i called my Xlite dialer, it gives me 484: address incomplete error message: While the enteries for extension in my table are like this: Table Enteries: Context exten priority appappdata Table data 1: test1_1234567 1 dial sip/192.168.0.200 Table data 2: test_12345 1 dial sip/192.168.0.197 Can you please guide me whats wrong here? Thanks ___ 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] incomplete address
When i add both of these extensions (12345 and 1234567) in the same context and try from CLI as : dial [EMAIL PROTECTED], asterisk recognize it. But it does not recognize the [EMAIL PROTECTED], asterisk only recognize 1234567#test1. Is this can be problem for the incomplete address error? Thanks On Fri, 2005-01-14 at 15:25, Muhammad Rizwan Khan wrote: Hello I have configured asterisk with getting user authentication from MySQL. Now the problem is when i called my Xlite dialer, it gives me 484: address incomplete error message: While the enteries for extension in my table are like this: Table Enteries: Context exten priority appappdata Table data 1: test1_1234567 1 dial sip/192.168.0.200 Table data 2: test_12345 1 dial sip/192.168.0.197 Can you please guide me whats wrong here? Thanks ___ 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] problem in calling
Hello I have configured asterisk on my lan, and using Xlite to call. When i called (e.g 12345) to the same extension (e.g 12345) it rings properly but when i called from (e.g 12345) to some other extension (e.g 123456) Xlite gives me error message that 484: address incomplete. Can you please guide me what to do here? Thanks ___ 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] Realtime / sip.conf
Hello Brian I am also trying to test realtime support for sip.conf, but having some problems during calls from one extensions to another.The problem which i am getting right now is that whenever i call from one extension (e.g 12345) to another (e.g. 123456), my dialler shows me error 484: address incomplete. Can you plz send me the data whcih you have inserted in mysql tables along with you sip.conf, extensions.conf and extconfig.conf files.(If you are not facing the problems like me). I 'll be really thankful to you. Thanks! On Fri, 2005-01-14 at 20:47, Brian S. Adelson wrote: I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see any debug messages on the console (sip debug). By just commenting out the sipfriends line in extconfig.conf the problem goes away. I do have the system utilizing realtime for Voicemail and Extensions, and I do not have any problems. Has anyone seen this problem before? extconfig.conf =-==-=-=-=-=-= [settings] realtime_ext = mysql,asterisk,extensions_table voicemail = mysql,asterisk,voicemail_table sipfriends = mysql,asterisk,sip_extensions *CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username asterisk for 10 minutes, 22 seconds. -Brian ___ 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] Realtime / sip.conf
I also have setting in extconfig.conf file, and i am able to register
users. The only difference is that i am using odbc instead of mysql in
settings.
The problem i am facing is that whenever i call (using Xlite) from one
extension (e.g 12345) to another (e.g. 123456), my dialler shows me
error 484: address incomplete. On the other hand whenever i call from
one extension (e.g 123245) to the same (e.g 12345) it grings properly.
Any idea what can be the problem here.
Thanks
On Fri, 2005-01-14 at 21:46, Brian S. Adelson wrote:
I do not recieve any debug messages for sip when extconfig is setup to
use sipfriends. Here is my extconfig:
[settings]
realtime_ext = mysql,asterisk,extensions_table
voicemail = mysql,asterisk,voicemail_table
sipfriends = mysql,asterisk,sip_extensions
As you can see, there is not much to it. But when I do have
sipfriends enabled, then I am not able to register any phones etc.
-Brian
On Fri, 14 Jan 2005 at 10:40 Matthew Boehm ([EMAIL PROTECTED]) wrote:
If you only get debug messages when you use console then you don't have
something setup right in extconfig
-Matthew
- Original Message -
From: Brian S. Adelson [EMAIL PROTECTED]
To: Matthew Boehm [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 14, 2005 10:16 AM
Subject: Re: [Asterisk-Users] Realtime / sip.conf
Sorry, thought I mentioned that.
In the debug, I do not see it attemping to query the mysql database.
It only makes this attempt when i try to pull information via the
console:
*CLI realtime load sipfriends name 155
Column Name Column Value
uniqueid 1
name 155
callerid X-Line Phone
canreinvite N
context from-internal
dtmfmode rfc2833
host dynamic
mailbox 155
nat no
port 5060
secret 155
type friend
username 155
regseconds 0
Jan 14 23:10:57 DEBUG[11834]: MySQL RealTime: Retrieve SQL: SELECT * FROM
sip_extensions WHERE name = '155'
Jan 14 23:10:57 DEBUG[11834]: MySQL RealTime: Everything is fine.
On Fri, 14 Jan 2005 at 10:12 Matthew Boehm ([EMAIL PROTECTED]) wrote:
What's in your debug?
-Matthew
- Original Message -
From: Brian S. Adelson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 14, 2005 9:47 AM
Subject: [Asterisk-Users] Realtime / sip.conf
I am currently in the process of testing out realtime support for
sip.conf. I have followed all of the directions that are listed in
the Wiki, but for some reason this does not work.
When utilizing a flat file, I am able to register endpoints without
any problems, and calls can proceed. One interesting side effect that
I have noticed is that when I am using realtime for sip, I am unable
to see any debug messages on the console (sip debug). By just
commenting out the sipfriends line in extconfig.conf the problem goes
away.
I do have the system utilizing realtime for Voicemail and Extensions,
and I do not have any problems. Has anyone seen this problem before?
extconfig.conf
=-==-=-=-=-=-=
[settings]
realtime_ext = mysql,asterisk,extensions_table
voicemail = mysql,asterisk,voicemail_table
sipfriends = mysql,asterisk,sip_extensions
*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username asterisk for
10
minutes, 22 seconds.
-Brian
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Re: [Asterisk-Users] Realtime / sip.conf
Brain: I am still hanging with the same problem, although i tried this: # iptables -t nat -A PREROUTING -p udp -i eth0 --dport 0 -j REDIRECT --to-port 5060 from: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20grandstream%20budgetone#comments But still have same problems? Any how are you able to make call now? On Fri, 2005-01-14 at 23:39, Brian S. Adelson wrote: Thank you everyone for your help. It looks like my problem was related to: http://bugs.digium.com/bug_view_page.php?bug_id=0003332 I patched and all is well now. -brian On Fri, 14 Jan 2005 at 12:57 Brian S. Adelson ([EMAIL PROTECTED]) wrote: Muhammad, Could you possible share you configuration and what version of asterisk you are running (I am using the head version from today) Maybe this will shed some light on the problem that both of us are having. -Brian On Fri, 14 Jan 2005 at 22:08 Muhammad Rizwan Khan ([EMAIL PROTECTED]) wrote: I also have setting in extconfig.conf file, and i am able to register users. The only difference is that i am using odbc instead of mysql in settings. The problem i am facing is that whenever i call (using Xlite) from one extension (e.g 12345) to another (e.g. 123456), my dialler shows me error 484: address incomplete. On the other hand whenever i call from one extension (e.g 123245) to the same (e.g 12345) it grings properly. Any idea what can be the problem here. Thanks On Fri, 2005-01-14 at 21:46, Brian S. Adelson wrote: I do not recieve any debug messages for sip when extconfig is setup to use sipfriends. Here is my extconfig: [settings] realtime_ext = mysql,asterisk,extensions_table voicemail = mysql,asterisk,voicemail_table sipfriends = mysql,asterisk,sip_extensions As you can see, there is not much to it. But when I do have sipfriends enabled, then I am not able to register any phones etc. -Brian On Fri, 14 Jan 2005 at 10:40 Matthew Boehm ([EMAIL PROTECTED]) wrote: If you only get debug messages when you use console then you don't have something setup right in extconfig -Matthew - Original Message - From: Brian S. Adelson [EMAIL PROTECTED] To: Matthew Boehm [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2005 10:16 AM Subject: Re: [Asterisk-Users] Realtime / sip.conf Sorry, thought I mentioned that. In the debug, I do not see it attemping to query the mysql database. It only makes this attempt when i try to pull information via the console: *CLI realtime load sipfriends name 155 Column Name Column Value uniqueid 1 name 155 callerid X-Line Phone canreinvite N context from-internal dtmfmode rfc2833 host dynamic mailbox 155 nat no port 5060 secret 155 type friend username 155 regseconds 0 Jan 14 23:10:57 DEBUG[11834]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_extensions WHERE name = '155' Jan 14 23:10:57 DEBUG[11834]: MySQL RealTime: Everything is fine. On Fri, 14 Jan 2005 at 10:12 Matthew Boehm ([EMAIL PROTECTED]) wrote: What's in your debug? -Matthew - Original Message - From: Brian S. Adelson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2005 9:47 AM Subject: [Asterisk-Users] Realtime / sip.conf I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see any debug messages on the console (sip debug). By just commenting out the sipfriends line in extconfig.conf the problem goes away. I do have the system utilizing realtime for Voicemail and Extensions
[Asterisk-Users] sound problem
I have configured asterisk, but when i calls from my dialler, it connects successfully, but did not give any voice at both ends. What should i need to do? Thanks ___ 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] sound problem
Is there any config file related to voice, which should be change in order to hear the sound in dialer? On Monday 10 January 2005 21:23, you wrote: I have configured asterisk, but when i calls from my dialler, it connects successfully, but did not give any voice at both ends. What should i need to do? Thanks ___ 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] Asterisk with MySQL
Please find the attached files, Thanks On Friday 07 January 2005 22:24, you wrote: post your /etc/odbc.ini and /etc/odbcinst.ini -matthew - Original Message - From: rizwan [EMAIL PROTECTED] To: Asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:19 AM Subject: [Asterisk-Users] Asterisk with MySQL Hello I am getting this error message, when i try to authenticate my users through database. Jan 7 20:28:08 WARNING[26487]: res_config_odbc.c:69 realtime_odbc: SQL Alloc Handle failed! Jan 7 20:28:08 NOTICE[26487]: chan_sip.c:7974 handle_request: Registration from 'rizwan sip:[EMAIL PROTECTED]' failed for '192.168.0.149' My conf files are: ;res_odbc.conf [test] dsn = test username = root password = pre-connect = yes ;extensions.conf [test] switch = Realtime/@realtime_ext ;extconfig.conf sipfriends = odbc,test,sip_buddies realtime_ext = odbc,test,extensions_table Can you please help me, what to do here? Thanks ___ 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 [FB_SAMPLE] Driver = FIREBIRD Description = Firebird driver Database= server:employee.gdb User= sysdba Password= masterkey With_Schema = 0 Dialect = 3 Charset = Role= Nowait = 0 OldMetaData = 0 ExecProc= 0 Dquote = 0 WithDefault = 1 TxnMode = 1 Flusfcommit = 0 Padvarchar = 0 Nullschema = 0 Fixprecision= 0 Simpleunicode = 0 wchardefault= 0 [demo] Driver = OOB Description = Easysoft ODBC-ODBC Bridge demo data source SERVER = demo.easysoft.com PORT= TRANSPORT = tcpip TARGETDSN = pubs LOGONUSER = demo LOGONAUTH = easysoft TargetUser = demo TargetAuth = easysoft [PostgreSQL] Description = ODBC for PostgreSQL Driver = /usr/lib/libodbcpsql.so Setup = /usr/lib/libodbcpsqlS.so FileUsage = 1 [FIREBIRD] Description = Easysoft Firebird ODBC Driver Driver = /usr/local/easysoft/fb/libfbodbc.so Setup = /usr/local/easysoft/fb/libfbodbcS.so FileUsage = 1 DontDLClose = 1 [OOB] Description = Easysoft ODBC-ODBC Bridge Driver = /usr/local/easysoft/oob/client/libesoobclient.so Setup = /usr/local/easysoft/oob/client/libesoobsetup.so FileUsage = 2 [MySQL ODBC 3.51 Driver] DRIVER = /usr/lib/libmyodbc3.so SETUP = /usr/lib/libmyodbc3S.so FileUsage = 1 ___ 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] Asterisk with MySQL
save biiling info into database instead of file. On Saturday 08 January 2005 04:04, you wrote: Hello Finally, i authenticated the user with database. and got this message. -- SIP Seeding '12345' at [EMAIL PROTECTED]:5060 for 1800 Special thanks to Mr. Matthew Boehm. Can you plz. guide me how i can configure prepaid with this setup. Thanking you again. ___ 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] Asterisk with MySQL
Hello
I am trying to configure asterisk userauthentication from database.
I am using Asterisk RealTime. http://www.voip-info.org/wiki-Asterisk+RealTime
But the problem is, whenver i try to call from Xlite (on my lan). It gave me
following error message.
Jan 7 00:20:22 NOTICE[15913]: chan_sip.c:7974 handle_request: Registration
from 'inam sip:[EMAIL PROTECTED]' failed for '192.168.0.197'
Enteries in database asterisk are as follows:
extensions_table: [table]
id context exten priority app appdata
Edit Delete 1 default 574555 1 Wait 2
Edit Delete 2 default 574555 2 SayNumber 102
Edit Delete 3 default 2815551212 1 Playback pbx-invalid
sip_buddies: [table]
uniqueid name accountcode amaflags callgroup callerid
1 12345 NULL NULL NULL 12345
canreinvite context defaultip
NULL default NULL
dtmfmode fromuser fromdomain host incominglimit
NULL NULL NULL dynamic NULL
outgoinglimit insecure language
NULL NULL NULL
mailbox md5secret nat permit deny pickupgroup port
NULL NULL NULL NULL NULL NULL 5060
qualify restrictcid rtptimeout
NULL NULL NULL
rtpholdtimeout secret type username allow disallow
NULL blah 12345 NULL NULL
regseconds ipaddr
100 192.168.0.197
For further details regadring my configurations, sip.conf,
res_odbc.conf, extconfig.conf, extensions.conf are attached with email.
Please help me, what i should do here to authenticate my users from MySQL
database.
Thanks;
; Static and realtime external configuration
; engine configuration
;
; Please read doc/README.extconfig for basic table
; formatting information.
;
[settings]
;
; Static configuration files:
;
; file.conf = driver,database[,table]
;
; maps a particular configuration file to the given
; database driver, database and table (or uses the
; name of the file as the table if not specified)
;
;uncomment to load queues.conf via the odbc engine.
;
;queues.conf = odbc,asterisk,ast_config
;
; Realtime configuration engine
;
; maps a particular family of realtime
; configuration to a given database driver,
; database and table (or uses the name of
; the family if the table is not specified
;
;iaxfriends = odbc,asterisk
;sipfriends = odbc,asterisk
;voicemail = odbc,asterisk
;extensions = odbc,asterisk
sipfriends = mysql,asterisk,customer_lines
voicemail = mysql,test
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=yes
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the include command that includes contexts
within
; other contexts. The #include command works in all asterisk configuration
files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than General and Globals represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N
Re: [Asterisk-Users] Asterisk with MySQL
Thanks for your reply,,, But still it does not work. Can you plz view my
attached files,,,
And how can i make sure that connection with the database successfully opened
or not?
Thanks and Regards!
On Friday 07 January 2005 01:57, you wrote:
Edit Delete 1 default 574555 1 Wait 2
Edit Delete 2 default 574555 2 SayNumber 102
Edit Delete 3 default 2815551212 1 Playback pbx-invalid
Lines 1 2 are missing _ for pattern matching.
rtpholdtimeout secret type username allow disallow
NULL blah 12345 NULL NULL
type doesnt seem to be either peer, user or friend.
-Matthew
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;
; Static and realtime external configuration
; engine configuration
;
; Please read doc/README.extconfig for basic table
; formatting information.
;
[settings]
;
; Static configuration files:
;
; file.conf = driver,database[,table]
;
; maps a particular configuration file to the given
; database driver, database and table (or uses the
; name of the file as the table if not specified)
;
;uncomment to load queues.conf via the odbc engine.
;
;queues.conf = odbc,asterisk,ast_config
;
; Realtime configuration engine
;
; maps a particular family of realtime
; configuration to a given database driver,
; database and table (or uses the name of
; the family if the table is not specified
;
;iaxfriends = odbc,asterisk
;sipfriends = odbc,asterisk
;voicemail = odbc,asterisk
;extensions = odbc,asterisk
sipfriends = mysql,asterisk,customer_lines
voicemail = mysql,test
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=yes
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the include command that includes contexts
within
; other contexts. The #include command works in all asterisk configuration
files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than General and Globals represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension. The priority
; next or n means the previous priority plus one, regardless of whether
; the previous priority was associated with the current extension or not.
; The priority same or s means the same as the previously specified
; priority, again regardless of whether the previous entry was for the
; same extension. Priorities may be immediately followed by a plus sign
; and another integer to add that amount (most
