[asterisk-users] ODBC and SQLIte3

Hi, I had configured Asterisk to use default database located in /var/lib/asterisk/sqlite3dir/sqlite3.db. When I put odbc show in Asterisk's cli, It returns me that I have conected but when I put sip show peers,Asterisk doesn't found any peer or user. ubuntu*CLI odbc show ODBC DSN Settings

Re: [asterisk-users] ODBC and SQLIte3

hi, if you use realtime peers, and you want to see their states, you have to look in the database... if you want to see their states via cli, you have to set rtcachefriends=yes in your sip.conf... there are other settings that you might be interested in... : rtcachefriends=yes ; Cache

Re: [asterisk-users] ODBC and SQLIte3

Hi, I have add this options into Sip.conf but the CLI continues telling the same message: ubuntu*CLI sip show peers Name/usernameHost Dyn Forcerport ACLPort Status Description Realtime 0

Re: [asterisk-users] ODBC and SQLIte3

looks like a mistake in your extconfig.conf... do you want to use realtime extensions too? for further instructions show us your extensions.conf and the verbose output of the cli showing the dialattempt... regards, yves Am 17.02.2013 14:31, schrieb termo termosel: Hi, I have add this

[asterisk-users] Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls

Hi everybody, We installed the Gigaset C610IP to one of our customer, those phone are natted and connects to our Asterisk 1.8.19. Each handset has is own account on our Asterisk, lets say Handset_102, Handset_103 and Handset_104. Problem is this one, taken from a sip show peers:

Re: [asterisk-users] Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls

On 17/2/13 5:02 pm, Administrator TOOTAI wrote: customer102/Handset_102 xxx.yyy.zzz.153 D N 5062 OK (80 ms) customer103/Handset_103 xxx.yyy.zzz.153 D N 5062 OK (70 ms) customer104/Handset_104 xxx.yyy.zzz.153 D

Re: [asterisk-users] ODBC and SQLIte3

Hi, tomorrow I will send you the extensions.conf and output cli when we call. Now, I have a message in cli, it appears alone without any input in the console. The warning is this: db.c:329 ast_db_put: Couldn't execute statment: SQL logic error or missing database Is it linked with my

[asterisk-users] Caller ID to be identical across all lines

I have two SIP lines, 09271 09974 , the 09271 is my publicly list phone line which is also my second pick, when i make an outgoing call , I use 09974, what I want is the caller ID for 09974 to be 09271 to the people I call. The reason is.. when i ring anyone i want them

[asterisk-users] asterisk-1.8 change in meetme behavior was this on purpose?

Hi. After the latest upgrade of asterisk-1.8, I notice that meetme does not allow the menu, if musiconhold is active which only occurrs if a single user is in the conference. Sometimes I have to unmute someone because for various reasons they cannot do this and they are the first one in the

Re: [asterisk-users] Caller ID to be identical across all lines

On Mon, 18 Feb 2013, Paul Edgar wrote:  I have tried to change the caller ID in the trunk routes there was no differnece, what i do at the moment is sent 09974 calls to the IVR of 09271 but it would be nice to have an elegant solution.  Is the caller ID generated at the phone exchange

[asterisk-users] Faxdetect + T38gateway

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm using faxdetect so that users can receive faxes on their phone numbers. It works fine. Fax is actually received by Hylafax through iaxmodem. I'm also using T.38 between asterisk and ATA (HT502/503) for better reliability. Sending from

Re: [asterisk-users] Faxdetect + T38gateway

On 18/02/2013 6:25 AM, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm using faxdetect so that users can receive faxes on their phone numbers. It works fine. Fax is actually received by Hylafax through iaxmodem. I'm also using T.38 between asterisk and ATA

[asterisk-users] ControlPlayback unable to play MixMonitor files, Premature EOF

I have a strange problem with MixMonitor and ControlPlayback, I am recording wav files MixMonitor(/path/filename.wav,b) When I try to play them back, I get no audio, dialplan continues to the next line as if the sound file is 0 seconds long ControlPlayback(/path/filename) I can play

Re: [asterisk-users] Faxdetect + T38gateway

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 17/02/2013 13:17, Larry Moore a ￩crit : You have not provided any information relating to your configurations. Thanks for your reply. I've tried your settings, no luck. I'll try to better describe my problem. T.38 from HT503 to Asterisk +

Re: [asterisk-users] confbridge and talker

11.02.2013 18:19, Matthew Jordan пишет: On 02/11/2013 01:20 AM, Dmitry Melekhov wrote: Hello! We use meetme, but, as I understand it will be soon removed from asterisk (already marked as deprecated), so I'm thinking about confbridge migration. Really, we use self-developed (really my ;-) )