Asterisk RealTime Sip
Asterisk RealTime Sip
Asterisk RealTime Sip
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Extconfig.conf Setup with Asterisk 1.6.1.1 Turnkey Provisioning at your data
center
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wish: 3CX Software PBX for
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sipusers => mysql,general,sip_buddies Auto Configures Phones & Trunks
Android, iOS, Windows & Mac
sippeers => mysql,general,sip_buddies clients
extensions => mysql,general,extensions_table *Rates shown do not include E-911 charges or government mandated taxes. Providers offering
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Database Config
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put the following in res_mysql.conf
Options to disconnect far
end when network loss
by alysonkeenan Wed 26 of
Oct, 2016 [general]
dbhost = 127.0.0.1
One way Audio dbname = asterisk
by profesorpenguin Tue 18
dbuser = myuser
of Oct, 2016
dbpass = mypass
ANSWER CALL FROM CLI dbport = 3306
CONSOLE
by giezireyes Wed 12 of Oct,
2016
Database Table
Lets create the table we need:
NOTE: You can use any table name you wish, just make sure the table name matches what you have the family name bound to.
NOTE: General principles: the column names in your database table correspond to the option names in sip.conf. You do not have to have all option
names defined in your table you only have to define those columns you actually use in sip.conf. Exceptions to this are 'regserver' and 'regseconds',
which the channel driver's realtime routines use for internal book-keeping. The 'name' field must also be present to hold equivalent of the [category
name] in the sip.conf file. It is easily possible that different versions of Asterisk will require different tables. For instance, a 1.6 version of Asterisk
would not use the "cancallforward", or 'restrictcid', or 'mask', or 'qualify', or "musiconhold" columns, but might require 'mohinterpret', 'mohsuggest',
etc. etc. options instead. Options (column names) that not offered in sip.conf are ignored. Some options in sip.conf are OK to have multiple entries,
but in a real-time database, Only one column is available. In these cases, you can add multiple values separated by semicolons. This occurs with
setvar, allow/disallow, and permit/deny.
NOTE: Column order is important!! If you place "ipaddr" before "host" (in the case of dynamic), you will never load the public IP address of your sip
device, as it will be overwritten when "host" is encountered. allow/disallow and permit/deny, the order of these statements is crucial in the config file,
as they are applied in order. In the realtime db, the order is determined by the order of the columns in the table. You will note that the deny/disallow
entries come before the allow/permit entries, to support the common usage of 'deny all', then permit '192.168.....'.
#
# Table structure for table `bit_sip_buddies`
#
Be sure that fields 'deny', 'permit', 'mask' contain valid IP or are set to "(NULL)" value (if no ACL control is needed)
be sure that field 'defaultip' is not null (set value to '0.0.0.0' if you don't need it).
(3/16/05) Updated by: utdrmac - incominglimit and outgoinglimit are deprecated. Use Asterisk cmd SetGroup instead.
NOTE: The index created on the column 'name' is because RealTime does its SELECT query using that column everytime. That column must also
be unique.
You do not need every column listed above. If you wish, you can remove those columns you know you will never use. The columns in your tables
should line up with the fields you would specify in the given entity declaration. If an entry would appear more than once, in the column it should be
separated by a semicolon. For example, an entity that looks like:
[foo]
host=dynamic
secret=bar
context=default
allow=gsm
allow=ulaw
You do not need to insert the ipaddr, port or regseconds information. These columns will be updated periodicaly by RealTime.
Testing
Throw some data into the above table and try to register an extension. The /var/log/asterisk/debug should give info on any problems.
Realtime Caching...
As of CVS-HEAD 3/16/05, if you enable RealTime caching in your sip.conf, Voicemail MWI works and so does 'sip show peers'. To do so, add
"rtcachefriends=yes" to the general section of your sip.conf file.
As the name implies, this caches the "RealTime" information from the database. As a result, there is a delay in updating some (if not all) fields in the
SIP entry when you update the database. For instance, if you create an entry with a context = "context1" and Asterisk loads it from the database
(perhaps the phone registered or tried to make a call), Asterisk holds on to that information as far as I can tell, indefinitely until a sip reload occurs.
This also means that you will have to do a sip reload to clear out any entries. Removing them from the database does not seem to work. You can still
add new entries though without reloading Asterisk.
RealTime caching...isn't that an oxymoron anyways? :-) (Someone check to see if this affects IAX too, I don't have any IAX phones at the moment.
- Flobi) (RealTime caching does affect IAX as well, works the same way. - Josh)
Update : use "sip prune realtime PEERNAME" then "sip show peer PEERNAME load" to flush the peer and reload from db - (Voicemeup)
Answer: No. "The templates are part of the configuration file (text files) parser and not supported in databases."
Reference: http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/276114
(Added 14 Sept 2013 bluecrow76)
See Also
Asterisk config sip.conf
Asterisk RealTime
Asterisk RealTime Static
Asterisk RealTime Sip
Asterisk RealTime IAX
Asterisk RealTime Voicemail
Asterisk RealTime Extensions
Created by: utdrmac, Last modification: Sat 14 of Sep, 2013 (10:06) by bluecrow76
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