Re: [asterisk-users] realtime config for general settings in sip.conf
contrib/realtime/ directory talks about sip peer/client parameters not general section(sip.conf) parameters like bindaddr, bindport, domain, realm, qualify etc... thanks, Kamlesh From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Thu, 3 May 2012 08:39:28 +0100 Subject: Re: [asterisk-users] realtime config for general settings in sip.conf You need 2 but they can point to the same table sipusers = sippeers = You can get table definitions by downloading the source and then looking in the contrib/realtime/ directory Ish On Thu, 2012-05-03 at 04:56 +, Kamlesh Kumar wrote: Hello, For realtime configuration, in /etc/asterisk/extconfig.conf file, what should be the family name to configure general sip.conf parameters. family name = driver,database name,table name thanks, Kamlesh From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Wed, 2 May 2012 13:59:58 +0100 Subject: Re: [asterisk-users] realtime config for general settings in sip.conf On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote: Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh Hi You can set defaults in the column definitions and you can still set globals in the sip.conf Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] realtime config for general settings in sip.conf
AFAIK, those setting need to be configured in the sip.conf which will still be parsed and acted upon even if you're using RTA On Fri, 2012-05-04 at 09:54 +, Kamlesh Kumar wrote: contrib/realtime/ directory talks about sip peer/client parameters not general section(sip.conf) parameters like bindaddr, bindport, domain, realm, qualify etc... thanks, Kamlesh From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Thu, 3 May 2012 08:39:28 +0100 Subject: Re: [asterisk-users] realtime config for general settings in sip.conf You need 2 but they can point to the same table sipusers = sippeers = You can get table definitions by downloading the source and then looking in the contrib/realtime/ directory Ish On Thu, 2012-05-03 at 04:56 +, Kamlesh Kumar wrote: Hello, For realtime configuration, in /etc/asterisk/extconfig.conf file, what should be the family name to configure general sip.conf parameters. family name = driver,database name,table name thanks, Kamlesh From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Wed, 2 May 2012 13:59:58 +0100 Subject: Re: [asterisk-users] realtime config for general settings in sip.conf On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote: Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh Hi You can set defaults in the column definitions and you can still set globals in the sip.conf Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] realtime config for general settings in sip.conf
You need 2 but they can point to the same table sipusers = sippeers = You can get table definitions by downloading the source and then looking in the contrib/realtime/ directory Ish On Thu, 2012-05-03 at 04:56 +, Kamlesh Kumar wrote: Hello, For realtime configuration, in /etc/asterisk/extconfig.conf file, what should be the family name to configure general sip.conf parameters. family name = driver,database name,table name thanks, Kamlesh From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Wed, 2 May 2012 13:59:58 +0100 Subject: Re: [asterisk-users] realtime config for general settings in sip.conf On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote: Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh Hi You can set defaults in the column definitions and you can still set globals in the sip.conf Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] realtime config for general settings in sip.conf
2012/5/2 Kamlesh Kumar kamlesh_...@hotmail.com Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh For what I have discovered, it is not possible. I hope to be wrong, but the sip.conf realtime is limited to peers (or users) registering on the box. It is not suitable even for defining trunks to be used by asterisk. Leandro -- _ -- 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] realtime config for general settings in sip.conf
On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote: Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh Hi You can set defaults in the column definitions and you can still set globals in the sip.conf Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] realtime config and extensions.conf
Meant to add that this is 1.4.26... :)
On Fri, 7 Aug 2009, Jeff LaCoursiere wrote:
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten = _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include = trunklocal
include = trunktollfree
[longdistance]
include = local
include = trunkld
[international]
include = longdistance
include = trunkint
[from-pstn]
exten = 7157999,1,VoicemailMain()
switch = Realtime
[residential]
include = from-pstn
include = international
---
And the relevant entries in the DB:
mysql select name, context from sip_buddies;
+-+-+
| name| context |
+-+-+
| 7157986 | residential |
| 7157980 | residential |
+-+-+
2 rows in set (0.01 sec)
mysql select * from extensions;
++-+-+--++-+
| id | context | exten | priority | app| appdata |
++-+-+--++-+
| 10 | residential | 7157986 |1 | Dial | SIP/7157986 |
| 11 | residential | 7157986 |2 | Congestion | |
| 12 | residential | 7157980 |1 | Dial | SIP/7157980 |
| 13 | residential | 7157980 |2 | Congestion | |
++-+-+--++-+
4 rows in set (0.00 sec)
---
The phone I am testing with has a sip entry in sip_buddies with a context
of residential. As you can see from the cascading contexts above the
residential context can dial local 7 digit numbers via the TRUNK (a zap T1
with an inbound context of from-pstn), but dialing the Voicemail main
number, also seven digits, overrides this and is executed directly. This all
works as expected and seems fairly elegant.
I also expected that the switch = Realtime statement in [from-pstn]
would allow any local numbers in the extensions table to also override the
trunk dialing, but it does not. So my test phone, when it dials a local
number that exists in the extensions table, ends up sending the call out
the TRUNK, then it comes back in the TRUNK on another channel, and then dials
the SIP phone as expected. The call at least goes through :) But it does
kill the video H.264 stream I was hoping for!
How can I make sure that the realtime entries override the pattern matching
in [trunk-local]?
Thanks,
j
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Re: [asterisk-users] realtime config and extensions.conf
On Fri, 7 Aug 2009, Jeff LaCoursiere wrote:
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten = _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include = trunklocal
include = trunktollfree
[longdistance]
include = local
include = trunkld
[international]
include = longdistance
include = trunkint
[from-pstn]
exten = 7157999,1,VoicemailMain()
switch = Realtime
[residential]
include = from-pstn
include = international
---
And the relevant entries in the DB:
mysql select name, context from sip_buddies;
+-+-+
| name| context |
+-+-+
| 7157986 | residential |
| 7157980 | residential |
+-+-+
2 rows in set (0.01 sec)
mysql select * from extensions;
++-+-+--++-+
| id | context | exten | priority | app| appdata |
++-+-+--++-+
| 10 | residential | 7157986 |1 | Dial | SIP/7157986 |
| 11 | residential | 7157986 |2 | Congestion | |
| 12 | residential | 7157980 |1 | Dial | SIP/7157980 |
| 13 | residential | 7157980 |2 | Congestion | |
++-+-+--++-+
4 rows in set (0.00 sec)
---
The phone I am testing with has a sip entry in sip_buddies with a context
of residential. As you can see from the cascading contexts above the
residential context can dial local 7 digit numbers via the TRUNK (a zap T1
with an inbound context of from-pstn), but dialing the Voicemail main
number, also seven digits, overrides this and is executed directly. This all
works as expected and seems fairly elegant.
I also expected that the switch = Realtime statement in [from-pstn]
would allow any local numbers in the extensions table to also override the
trunk dialing, but it does not. So my test phone, when it dials a local
number that exists in the extensions table, ends up sending the call out
the TRUNK, then it comes back in the TRUNK on another channel, and then dials
the SIP phone as expected. The call at least goes through :) But it does
kill the video H.264 stream I was hoping for!
How can I make sure that the realtime entries override the pattern matching
in [trunk-local]?
Thanks,
j
And now to answer my own silly question...
The switch statement will use the static context it is a member of to
search the tables, and I had 'residential' rather than 'from-pstn' in the
tables.
Works fine now :)
Cheers,
j
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Re: [Asterisk-Users] Realtime Config
As far as I can tell with our systems, the config files are read first, then the realtime db. We've got a few static servers that never change, so I hardcode those in case something goes wrong with the DB, and the DB contains any other configurations that will be dynamic. I'm not sure if realtime has any support for the basic general information at the top of the config files, so I think you need to have the files to convey that information. Aaron Douglas Garstang wrote: I'm a little confused about something with Realtime. It isn't clear to me what order Asterisk prefers to read the config. If we are using realtime, do we have to completely throw away the use of the .conf files? Sometimes not it appears. Extensions.conf lets you have a switch command to call into Realtime. For other conf files, you can use the realtime static table to load the general sections, or can you? I guess this question doesn't make much sense because the docs don't make much sense to me. My preference is to have static stuff in the config files and have dynamic portions, ie bits that might change, in realtime. Doug. ___ --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] Realtime Config
Hi Aaron. Well, there's 'realtime static' which it supposedly uses. It's table structure is: CREATE TABLE `ast_config` ( `id` int(11) NOT NULL auto_increment, `cat_metric` int(11) NOT NULL default '0', `var_metric` int(11) NOT NULL default '0', `commented` int(11) NOT NULL default '0', `filename` varchar(128) NOT NULL default '', `category` varchar(128) NOT NULL default 'default', `var_name` varchar(128) NOT NULL default '', `var_val` varchar(128) NOT NULL default '', PRIMARY KEY (`id`), KEY `filename_comment` (`filename`,`commented`) ) TYPE=MyISAM; and you can use it to store information in the [general] section and so on. I know this works because I've used it before. It just isn't clear if all the config files use it or not. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 9:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime Config As far as I can tell with our systems, the config files are read first, then the realtime db. We've got a few static servers that never change, so I hardcode those in case something goes wrong with the DB, and the DB contains any other configurations that will be dynamic. I'm not sure if realtime has any support for the basic general information at the top of the config files, so I think you need to have the files to convey that information. Aaron Douglas Garstang wrote: I'm a little confused about something with Realtime. It isn't clear to me what order Asterisk prefers to read the config. If we are using realtime, do we have to completely throw away the use of the .conf files? Sometimes not it appears. Extensions.conf lets you have a switch command to call into Realtime. For other conf files, you can use the realtime static table to load the general sections, or can you? I guess this question doesn't make much sense because the docs don't make much sense to me. My preference is to have static stuff in the config files and have dynamic portions, ie bits that might change, in realtime. Doug. ___ --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 ___ --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] Realtime config
I got the CVS head to compile finally, and yes I ditched odbc. noob or not, it's a pain in the a$$ if you mess up the install. All in all, mysql seems to work fine. Thanks. Matt -Original Message- From: Joe Dennick [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 15, 2005 1:20 PM To: Asterisk Users Mailing List -Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime config Have you considered using the mysql method instead of the odbc method. I'm using it and it works just fine. Here's a sample of my extconfig.conf: extensions = mysql,ast-conf,extension sipfriends = mysql,ast-conf,sip_buddi voicemail = mysql,ast-conf,voicemail You also need to add the floowing to your res_mysql.conf file: [general] dbhost = 192.168.1.7 dbname = ast-conf dbuser = dbusername dbpass = blah dbport = 3306 dbsock = /tmp/mysql.sock The only two things I have found that doesn't work is a) the mailbox entry for a SIP user doesn't actually light up the MWI (Message Waiting Indicator); and b) voicemail passwords cannot begin with a '0' (zero) because its a numeric field. Matt Schulte ([EMAIL PROTECTED]) wrote: Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without saying too much more here's the configs: odbcinst.ini [mysql] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc3.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 what I can see in the console. It shows no queries or anything, driving me nuts. I'm running asterisk 1.0.6, as head won't seem to compile (as of this this email).. I'm trying to test realtime via simply SIP REGISTER: Mar 15 13:40:39 NOTICE[7905]: chan_iax2.c:3910 register_verify: No registration for peer 'brak-test' (from blah blah) Mar 15 13:40:39 NOTICE[7906]: chan_sip.c:7681 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for 'blah' ___ 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 -- Joe Dennick [EMAIL PROTECTED] ___ 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 config
Have you considered using the mysql method instead of the odbc method. I'm using it and it works just fine. Here's a sample of my extconfig.conf: extensions = mysql,ast-conf,extension sipfriends = mysql,ast-conf,sip_buddi voicemail = mysql,ast-conf,voicemail You also need to add the floowing to your res_mysql.conf file: [general] dbhost = 192.168.1.7 dbname = ast-conf dbuser = dbusername dbpass = blah dbport = 3306 dbsock = /tmp/mysql.sock The only two things I have found that doesn't work is a) the mailbox entry for a SIP user doesn't actually light up the MWI (Message Waiting Indicator); and b) voicemail passwords cannot begin with a '0' (zero) because its a numeric field. Matt Schulte ([EMAIL PROTECTED]) wrote: Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without saying too much more here's the configs: odbcinst.ini [mysql] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc3.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 what I can see in the console. It shows no queries or anything, driving me nuts. I'm running asterisk 1.0.6, as head won't seem to compile (as of this this email).. I'm trying to test realtime via simply SIP REGISTER: Mar 15 13:40:39 NOTICE[7905]: chan_iax2.c:3910 register_verify: No registration for peer 'brak-test' (from blah blah) Mar 15 13:40:39 NOTICE[7906]: chan_sip.c:7681 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for 'blah' ___ 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 -- Joe Dennick [EMAIL PROTECTED] ___ 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 config
Matt Schulte wrote: anything, driving me nuts. I'm running asterisk 1.0.6, as head won't Take your 'blah-blah' to the 'blah-blahtologist'. - Dr. Cox, Scrubs RealTime requires CVS-HEAD! That is why its not working with 1.0.6!!! Perhaps I should make the font on the wiki larger..hmm.. -Matthew ___ 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 config
The only two things I have found that doesn't work is a) the mailbox entry for a SIP user doesn't actually light up the MWI (Message Waiting Indicator); and b) voicemail passwords cannot begin with a '0' (zero) because its a numeric field. You are behind the times. The MWI now works. You need to checkout the newest sip.conf.sample and look for rtsipcache (or something like that). You can simply change the password field from INT (4) to VARCHAR(4). Lets also be clear that problem a. with the MWI is NOT a res_config_mysql issue. It is a RealTime Issue and not specific to any ARA driver. -Matthew ___ 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
