search for: sip_users

...ing. Furthermore, cdr_odbc is working, and actively logging my failed call attempts to db through ODBC using the same DSN. unixODBC and the mysql drivers are installed from source. Here are the relevant parts of the config: Extconfig.conf (Under the [settings] section) sipusers => odbc,voip,sip_users sippeers => odbc,voip,sip_users voicemail => odbc,voip,voicemail_users extensions => odbc,voip,extensions_table res_odbc.conf [asterisk] dsn => MySQL-asterisk username => voip password => temp123 pre-connect => yes Under extensions.conf, in the [from-sip] context: switch =&...

...myodbc3.so Setup = /usr/lib/libodbcmyS.so FileUsage = 1 odbc.ini --- Description = Asterisk MySQL Connection Trace = off TraceFile = stderr Driver = mysql Server = blah.blah User = blah Password = blah port = 3306 database = asterisk extconfig.conf iaxfriends => odbc,asterisk,sip_users sipfriends => odbc,asterisk,sip_users sipusers => odbc,asterisk,sip_users sippeers => odbc,asterisk,sip_users [asterisk] dsn => asterisk username => dffjdg password => blajh pre-connect => yes Ok, now that's out of the way. In my debug log it shows -nothing-, besides wh...

Hi all, we use OpenSER together with Asterisk. All SIP users registers with OpenSER and asterisk is doing the voicemail thing. We use the Asterisk RealtimeArchitecture for voicemail users and SIP peers. The database table for the sip peers is a view from the OpenSER subscriber table. The MWI for a user will only work, if the user object (sip peer) is loaded into memory and visible with the CLI

...tered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = 'brak-test' Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Everything is fine. Mar 24 09:16:47 NOTICE[4527]: Unable to create channel of type 'IAX2' (cause 3) Mar 24 09:16:47 VERBOSE[4527]: == Everyone is busy/congested at this time (1:0/1/0) Mar 24 09:16:47 DEBUG[4527]...

...p a SIP users/peers table in a mysql database and I have populated it with a few peers. I have configured asterisk addons and from the asterisk CLI I am able to search the sip users / peers tables using the "realtime load" command. This is after i added "sipusers => mysql,asterisk,sip_users" to my extconfig.conf file. However I don't know what to do to get asterisk to look at that table when a request to register comes from a sip peer. I understand that sipusers and sip peer are contradictory but they are all defined as peers. -------------- next part -------------- An HTML a...

...NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. I'd like to confirm the layout of the sip_users table in my MySQL database to make sure it's what Asterisk is looking for. I cannot find any official documentation that specifies what the sip_users table should look like. Their documentation system does a great job of showing what the table should look like for the Voicemail ODBC storage, fo...

Hi, I'm testing Asterisk with MySql and I would want to insert sip users in a table "sip_users". After I modified extconfig.conf with "sipusers => odbc,asterisk" and I create the table sipusers, which changes must I make to sip.conf? Thank's Maury P.S.: C'? qualche utente italiano nella mailing list? -------------- next part -------------- An HTML attachment was s...

...eta 5. I'm using realtime to store extensions and sip users. When a user logs on, everything works fine - calls are also fine. But I keep getting these in the console: [Mar 9 16:31:58] WARNING[10284]: res_config_pgsql.c:195 realtime_pgsql: PostgreSQL RealTime: Could not find any rows in table sip_users. Full log: http://rafb.net/p/LO7DEM95.html I think this part is a bit wierd: Query: SELECT * FROM sip_users WHERE name = '' Is this a bug or a known problem? I don't know where to even begin looking. Again, everything seems to be in working order - it's just an annoyance. Greeti...

...=no externip=10.100.1.31 extensions.conf [internal] switch => Realtime/internal at extensions res_mysql.conf [general] dbhost = 10.100.1.32 dbname = asterisk dbuser = asterisk dbpass = *********** dbport = 3306 dbsock = /var/lib/mysql/mysql.sock extconfig.conf sipusers => mysql,asterisk,sip_users sippeers => mysql,asterisk,sip_users extensions => mysql,asterisk,extensions_table And heres the DB config.... mysql> select * from extensions_table; +----+----------+-------+----------+--------+-----------+ | id | context | exten | priority | app | appdata | +----+----------+-----...

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hi if someone cares to add the solution to the bug tracker, others may find it... bkw's suggestion about enabling SIP_USERS worked perfectly thanks roy

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first rings at my phone and in the second i can see the callers number. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

...ariables with optional prefix to the name. e.g. prefix of 'var_' would make the column 'name' become the variable ${var_name} eg. extensions.conf (or in a db table) (lets assume that we have in our table column names: name username password phone_nbr context table is "sip_users" connected to familyname sipusers , there is a user with name=guru) [init] exten => s,1,Realtime(sipusers|name|guru) exten => s,2,Noop(${name}) ;gives use what we already know exten => s,3,Noop(${username}) ; guru username etc. simple , isnt it ? I hope it will save...

I have a CISCO 7912 phone, the LED on the phone does not glow when there is new voicemail, can we configure Asterisk to have the LED glow on new Voicemail. Regards, Sanjay Rajdev

...terisk/extensions: The logic to check whether the value of "regsever" is different from "sysname" and if so - redirect the call. The code fragments are (I am using AEL): To get the regserver from the database: MYSQL(Query resID ${connid} SELECT regserver from sip_users where name='${EXTEN}'); MYSQL(Fetch FetchId ${resID} RegServer); MYSQL(Clear ${resID}); so now RegServer contains the server where the phone is registered. Next: if(("${DEVSTATE(SIP/${EXTEN})}" == "UNAVAILABLE") ||...

...at iaxtermination/01115141234123 -- doing lookup for '195.66.85.55' -- Hungup 'IAX2/iaxtermination-2' == Spawn extension (autorized, *9215141234123, 1) exited non-zero on 'SIP/FJST1001-087b8138' -- doing lookup for '195.66.85.55' I know iax_users and sip_users on realtime are suposte to be dynamic, on an incomming request, ARA will look on to the database and pull the information necessary to authenticate the user and let them place calls, etc, and then after hangup ARA will delete/remove the information about that particular user, but this is for incomm...

Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...;' failed for '192.168.0.25' - Wrong password [Jul 21 15:40:47] DEBUG[2105]: chan_sip.c:15372 sipsock_read: SIP message could not be handled, bad request: 1832465624 at 192.168.0.25 This error is different to the error that is received if a username that is not in the MySQL sip_peers / sip_users table is specified. Therefore at least the MySQL connection appears to be working. extconfig.conf: sipusers => mysql,asterisk_config sippeers => mysql,asterisk_config I have also tried explicitly adding ',sip_users' and ',sip_peers' to these lines, but asterisk behaved simil...

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson