Known as Virtual Hold, youll have to program inside asterisk to achieve that. El 31/05/12 10:48, equis software escribió: > Is there any option in Asterisk distribution of this? > > Thanks. > > > — > __________________________________________________________________..
Will ODBC become the default then ? I see no ODBC-command to use in the dialplan. Jonas. On 05/05/2012 11:12 AM, Leandro Dardini wrote: > Use ODBC. Check the func_odbc.conf configuration file. > > Leandro > > 2012/5/5 Jonas Kellens> > > > > I not..
Use ODBC. Check the func_odbc.conf configuration file. Leandro 2012/5/5 Jonas Kellens > ** > > > I notice when upgrading from 1.6.2 to 1.8 that in the menuselect > app_mysql is indicated as deprecated. > > If one wants to use the MySQL-command in ..
Use local channel 2012/1/31 Niccolò Belli : > > Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to > call SIP/$TRUNK instead. > > Cheers, > Darkbasic > > — > __________________________________________________________________..
Would ExtensionStatus provide the data your looking for? Event: ExtensionStatus Privilege: call,all Exten: 216 Context: ext-local Hint: SIP/216 Status: 0 On Thu, Oct 6, 2011 at 9:12 AM, Olivierwrote: > > > I would like to receive an AMI event whene..
Cool thx 🙂 dont know why i didnt found it myself 😀 Quoting Patrick Lists : > On 07/10/2011 05:02 PM, Matiss Jekabsons wrote: >> Is there some detailed documentation for 1.8.5? I am tryin to make >> Asterisk 1.8.5 with MySQL backend, TLS transport ..
Set queue_log = no in logger.conf. By default it is set to yes. [SATISH] On Wed, Jun 8, 2011 at 12:30 PM, Jonas Kellens wrote: >list, > > I have configured extconfig.conf to save queue log into my MySQL-DB. > > I notice however that there is still logg..
On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote: > > > Im using SIP realtime with MySQL DB. > > Is it possible to get the status of the SIP peer (free / calling) from > this realtime DB ? > > If not, is there another way to obtain the call st..
how did you increase it? Am 16.02.2011 um 00:11 schrieb Hans Witvliet : > On Tue, 2011-02-15 at 18:06 +0100, Felix Dong wrote: >> >> >> >> could I adjust the Rx and Tx gains for SIP and CAPI? If it is >> possible, how should I do it? >> Thanks a l..
Can you post your confs. 2011/1/17 Arie Goldfeld > Hello! > > I have compiled Asterisk 18.104.22.168 on my SheevaPlug. It works all right for > me, except for one problem that I have encountered: I can only register a > SIP client (X-Lite in my case) if ..