Whatever you do, you do at your own risk. I can only recommend and do not claim 100% of the solution much depends on your environment and other settings. I can not guess. Addition of materials and fixing bugs is welcomed
Small notes on the topic of voice transmission or in other words the phones :)
- Also available: RU
Stuff happens, and sometimes it is necessary to perform a stop or restart the PBX after completion of all call or in a free moment. How to do it look below:
- Also available: RU
Debug SIP Protocol in Asterisk
If you need to see SIP messages coming from the peer, and what messages your server sends to it, then on the server you need to run the following command: "asterisk -rx "sip set debug peer SIPuserID"" SIPuserID, respectively, change the name of the SIP user account. It is important to take into account that the Asterisk server must know the user's IP, in other words, the user must be registered or have an IP registered in its settings.
- Also available: RU
GotoIfTime() - Performs a conditional transition based on time and day.
GotoIfTime(time, days of the week, days of the month,months?label)
Performs a queue or number transition if the time matches the time specified in the condition.
If you helped the article or information was useful. Gratitude should not know borders
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