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guides:voip [2010/02/21 16:00] fishy |
guides:voip [2010/02/22 00:30] (current) fishy |
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| Asterisk-generated audio due to bad timing etc. | Asterisk-generated audio due to bad timing etc. | ||
| - | - Update your system! If you install AsteriskNOW, and don't perform a "yum upgrade", you will be running a version of the dahdi-dummy module that depends on a hardware RTC. Xen does not have a hardware RTC. This is fixed in CentOS/Asterisk packages available using yum, so simply update. | + | - Update your system! If you install AsteriskNOW, and don't perform a "yum upgrade", you will be running a version of the dahdi-dummy module that depends on a hardware RTC. Xen does not have a hardware RTC. This is fixed in CentOS/Asterisk packages available using yum, so simply update. <code>yum upgrade</code> |
| - | - Edit the /etc/dahdi/modules file to not load _any_ modules. This will make sure only the dahdi_dummy module gets loaded. | + | - Edit the /etc/dahdi/modules file to not load _any_ modules. This will make sure only the dahdi_dummy module gets loaded.<code> |
| + | cp /etc/dahdi/modules /etc/dahdi/modules_dist | ||
| + | echo "# Empty" > /etc/dahdi/modules | ||
| + | </code> | ||
| - Edit /etc/asterisk/asterisk.conf, and add the following to the Options block <code>internal_timing=yes</code> | - Edit /etc/asterisk/asterisk.conf, and add the following to the Options block <code>internal_timing=yes</code> | ||
| - Restart your system. | - Restart your system. | ||
| + | |||
| + | ===== Meetme/Conferences not working? ===== | ||
| + | |||
| + | The symptom of this problem is: you have properly configured a Conference, but dialing the conference number | ||
| + | from an internal numer (i.e. extension), the Asterisk woman tells you: | ||
| + | <code> | ||
| + | That is not a valid conference number | ||
| + | </code> | ||
| + | |||
| + | Conferences requires proper generated timing, and the solution to this problem is the same as | ||
| + | the solution to MoH/ringtone problems above. Update your dahdi-modules and instruct asterisk to generate | ||
| + | internal timing from dahdi-dummy. | ||
| ===== No CDR-info? Reports are empty you say?===== | ===== No CDR-info? Reports are empty you say?===== | ||
| Line 76: | Line 91: | ||
| Look for the variable FOPPASSWORD.... | Look for the variable FOPPASSWORD.... | ||
| + | |||
| + | ===== Using Nortel telephones, is it possible? ===== | ||
| + | |||
| + | YES! Nortel proprietary IP telephones uses a communication protocol called Unistim. This protocol has been reverse-engineered for the purpose of interoperabillity (thus legal at least in Europe), and the results are implemented in the chan_unistim Asterisk module. I separated this in to its own page: | ||
| + | |||
| + | * [[guides:voip:unistim|Using Nortel Unistim phones with FreePBX/Asterisk/AsteriskNOW]] | ||
| + | |||
| + | At least the i2004, i2002, i2007 hardphones and the i2050 softphone are supported. | ||
| + | |||
| + | |||