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====== Some VoIP-stuff ;) ====== ===== SIP trunk settings ===== I am using inbound/outbound SIP with two Norwegian providers at the moment: * [[guides:voip:televoip|SIP trunk settings for TeleVoiP]] * [[guides:voip:phonzo|SIP trunk settings for Phonzo]] ===== Unable to upload MoH data? ===== I prefer uploading MP3-based MoH, and let freePBX handle the conversion to a "usable" format, as well as entering the file into any config that is needed. But, two things prevents this from working. First, the directory freePBX wants to use does not exist on an AsteriskNOW installation. So we need to create it... <code> mkdir -p /var/lib/asterisk/mohmp3/ chown asterisk:asterisk /var/lib/asterisk/mohmp3/ chmod +w /var/lib/asterisk/mohmp3/ </code> Second, the tools needed for the actual conversion is not installed! This may be because the tool used, mpg321/123 is not in CentOS repo's, and thus, it cannot be pulled in as a dependency. Manual installation is needed: <code> wget wget wget yum install libao rpm -Uhv libmad-0.15.1b-4.el5.rf.i386.rpm rpm -Uhv mpg321-0.2.10-8.el5.rf.i386.rpm rpm -Uhv libid3tag-0.15.1b-3.el5.rf.i386.rpm </code> ===== No MoH and choppy dial/ring when running on Xen? ===== I had lots of trouble with instabillity of the DAHDI (previously Zaptel) software, and decided to use pure simulated timing by disabling the DAHDI altogether. That is not a good idea, as it breaks any 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. <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.<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> - 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?===== Of some odd reason, AsteriskNOW ships with all features of FreePBX available, but no database-integration of CDR-data installed. So to get your CDR-data, you need to install asterisk addons to allow MySQL storage of data. <code> yum install asterisk-addons-mysql asterisk-addons-core asterisk-addons </code> <code> asterisk -rx "restart now" </code> To verify, you can do: <code> asterisk -rc *CLI> module show like mysql Module Description Use Count Simple Mysql Interface 0 MySQL CDR Backend 0 </code> Do not worry if more modules are listed, I just extracted the relevant ones... ===== What is the password for FOP? ===== Check, and update /etc/amportal.conf 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.