[asterisk-users] Asterisk 13.11 realtime problem registering phones

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general@ps_contacts: column 'qualify_timeout' cannot be

Re: [asterisk-users] Feature Request: what about "core stop panic" ?

On Fri, Sep 2, 2016 at 9:34 AM, Olivier wrote: > Hello, > > I had a recent case where Asterisk stopped due to a segfault. > This reminded me that being sure that whenever such issue occurs, it's > useful to have a core file or various data at hand to analyze and exchange >

[asterisk-users] Feature Request: what about "core stop panic" ?

Hello, I had a recent case where Asterisk stopped due to a segfault. This reminded me that being sure that whenever such issue occurs, it's useful to have a core file or various data at hand to analyze and exchange with support teams. How can you double check a running Asterisk system would

Re: [asterisk-users] voicemail greeting

> hi.i managed to record my voicemail greeting. the only problem is that after > my greeting the caller hear '...please leave your message after the tone. > when done press the pound key or hangup.' is there a way to get rid of that? > Ideally i would like to have my own recording and then the

[asterisk-users] voicemail greeting

hi.i managed to record my voicemail greeting. the only problem is that after my greeting the caller hear '...please leave your message after the tone. when done press the pound key or hangup.' is there a way to get rid of that? Ideally i would like to have my own recording and then the beep

Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

Le 02/09/2016 à 11:26, Jonas Kellens a écrit : Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now

[asterisk-users] How to disable subsequent transfers?

Hi, Consider the following scenario. A customer's incoming call enters the system, and after some processing, the call is placed on a queue, where it will be picked up by an agent. Then, the agent makes an attended transfer (using asterisk internal transfer) of this costumer to some other

[asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding