similar to: Realtime does not work yet, ...

Displaying 20 results from an estimated 20000 matches similar to: "Realtime does not work yet, ..."

2006 Jan 29
2
username not stabled?
vpbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 621/621 192.168.250.76 D N 5060 OK (65 ms) 626/626 192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified) D N 0 UNKNOWN 615/Ronald office 192.168.250.103 D N 5060 OK (41
2006 Feb 23
1
mysql problems
My database machine is broken and I have to use another one. I made somewhere mistake(s) and get now in the debug file: [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query: SELECT * FROM sip_buddies WHERE name = '886' [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query Failed because: Can't find file: './astconf/sip_buddies.frm' (errno: 13) [Feb 24 09:05:25]
2005 Mar 17
1
Realtime - how to reload ?
I had the impression that the command: *CLI> realtime load sippeers name 621 (The new configuration was displayed after that command) would re-load the config of phone 621 (I changed the context and tried above command, however, it kept the old info!) bye Ronald
2005 Oct 16
2
No voice - one way - both ways
I got four phones: 601 is a SIP phone (no brand) 615 is Snom 190 621 is a Grand stream 628 is a remote SIP phone (no brand) 601, 615, 628 can call each other without any problems 621 used to be able to call remote 628, but after upgrade to CVS Head Nov. 11 the remote party cannot hear me. 615 never could call remote 628, both party hear nothing. 601 can always call 628 [Oct 16 00:52:13] --
2009 Mar 02
2
Asterisk realtime
Hi all, I'm using asterisk in real time mode...All extensions are defined in table sip_buddies...Everything looks fine and asterisk is reading extensions info from the sip_buddies table...The problem occurs as soon as any information on an extension is changed from sip_buddies table...Which mean, if I change the secret field in sip_buddies table then i should reload asterisk to read again the
2010 Feb 22
2
Problems with SIP realtime
I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from '<sip:testsip at 192.168.1.150;transport=UDP>' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing
2010 Jun 08
6
reloading realtime sip peers
Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip
2007 Sep 26
4
Asterisk realtime error
Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:
2009 Dec 11
2
sip realtime question
Hi everybody, First of all i am sorry my English :) i want to configure my asterisk server as a sip server that stores sip users in the mysql database connecting directly over odbc driver. My odbc configuration works as below [root at ao042 asterisk]# isql -v asterisk +---------------------------------------+ | Connected! | | | |
2005 May 04
4
Problem with realtime SIP
Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons. I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages
2006 Jun 04
3
How to make this into a Macro?
I have for each phone such a paragraph in my dialplan. I would like to save this by using a Macro. How can I do that? exten => 8863959,1,Dial(SIP/8863959,60,r) exten => 8863959,2,NoOp(${DIALSTATUS}) exten => 8863959,3,Voicemail,u8863959@Customers exten => 8863959,104,Voicemail,b8863959@Customers exten => 8863959,105,hangup
2008 May 05
3
simple realtime question
HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango
2006 Jun 28
1
Realtime: how to use column setvar?
How can I use the column "setvar" in my dialplan? I am not sure if it is for that what I need: Many phones have the same jump in place, but need a few variables different, like tariff, silent, need_password, I have for tariff = 4 variations, for silent=2, for need_password=2 ... If I solve it like now, I need 4x4x2 = 32 context variations. If I could use a field in the Real-time
2009 Aug 07
2
realtime config and extensions.conf
Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include
2005 Aug 24
2
RealTime ignoringswitch=>Realtime/context@re altime_ext
Thanks John, You are my savior. This is such a great relief. Apparently realtime will not use either '127.0.0.1' or 'localhost' to connect to the database. I had to use the actual IP address attached to the NIC before it worked. My OS is Debian just a note and Asterisk HEAD from August 20, 2005 Details below for those who might be swimming in the same pool with me.
2006 Apr 04
1
voipstunt: "Forbidden - wrong password ..."
voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing
2006 Apr 10
1
RE: still no solution for me, if one provider
>Our user places a call, the gateway responds with no sound at all, or >hangs up, or gives busy tone. > >How can we get to the next provider? > >I have now: >exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-a) >;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-b) >;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-c) >exten =>
2010 Aug 03
1
sip.conf register in realtime DB
Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config
2005 Mar 24
1
realtime - unable to find key
ok so my table looks like this... REATE TABLE `sip` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip`
2005 Aug 20
2
Realtime sip_buddies "register=>" how?
Hi all I've been doing some testing on realtime using mysql, an have a little question that could not find the answer to or maybe its not posible at this time. Is there a way use "register=>......" on a DB using realtime. For the moment I use it in sip.conf. It will help me a lot if this could be store on a DB somehow. commets or sugestions .... ? thanks Billy