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guides:voip [2010/02/21 14:57] fishy created |
guides:voip [2010/02/22 00:30] (current) fishy |
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| ====== Some VoIP-stuff ;) ====== | ====== Some VoIP-stuff ;) ====== | ||
| - | ===== TeleVoiP SIP trunk settings ===== | + | ===== SIP trunk settings ===== |
| - | Asterisk/FreePBX/AsteriskNOW settings for working SIP-trunking | + | I am using inbound/outbound SIP with two Norwegian providers at the moment: |
| - | both inbound and outbound with TeleVoiP in Norway.. | + | |
| - | Replace 87654321 with your TeleVoiP account number/phone number, | + | * [[guides:voip:televoip|SIP trunk settings for TeleVoiP]] |
| - | and PassW0Rd with the associated password as given by TeleVoiP. | + | * [[guides:voip:phonzo|SIP trunk settings for Phonzo]] |
| - | <code> | + | ===== Unable to upload MoH data? ===== |
| - | Outbound Caller ID: 87654321 | + | |
| - | + | ||
| - | Trunk Name: 87654321 | + | |
| - | PEER Details: | + | |
| - | + | ||
| - | username=87654321 | + | |
| - | type=peer | + | |
| - | secret=PassW0Rd | + | |
| - | insecure=very | + | |
| - | host=sip.televoip.no | + | |
| - | fromuser=87654321 | + | |
| - | fromdomain=sip.televoip.no | + | |
| - | canreinvite=no | + | |
| - | + | ||
| - | USER Context: from-trunkN | + | |
| - | USER Details: | + | |
| - | + | ||
| - | type=user | + | |
| - | secret=PassW0Rd | + | |
| - | context=from-pstn | + | |
| - | + | ||
| - | Register String: | + | |
| - | 87654321:PassW0Rd@sip.televoip.no/87654321 | + | |
| - | </code> | + | |
| - | + | ||
| - | * fromdomain is required to make asterisk send data that TeleVoiP undestands on outgoing calls | + | |
| - | * canreinvite=no is essential for outbound audio on outbound calls | + | |
| - | * Adding the DID on the register string is essential to have inbound DID work in your dial-plans | + | |
| - | + | ||
| - | The DID, Direct Inward Dialing Number, is required if you plan to have more than one trunk, and/or | + | |
| - | more than one inbound route. When not setting the "/DID" part of the register-string with TeleVoiP, | + | |
| - | the do not give you anything useful on the DID, whereas if you set it, TeleVoiP will send you | + | |
| - | whatever you enter as a DID.... | + | |
| - | + | ||
| - | ==== 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. | 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. | ||
| Line 72: | Line 36: | ||
| </code> | </code> | ||
| - | ==== No MoH and choppy dial/ring when running on Xen? ==== | + | ===== 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 | I had lots of trouble with instabillity of the DAHDI (previously Zaptel) software, and decided to use | ||
| Line 78: | Line 42: | ||
| 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. | ||
| - | ==== No CDR-info? Reports are empty you say?==== | + | ===== 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, | Of some odd reason, AsteriskNOW ships with all features of FreePBX available, | ||
| Line 107: | Line 86: | ||
| Do not worry if more modules are listed, I just extracted the relevant ones... | Do not worry if more modules are listed, I just extracted the relevant ones... | ||
| - | ==== What is the password for FOP? ==== | + | ===== What is the password for FOP? ===== |
| Check, and update /etc/amportal.conf | Check, and update /etc/amportal.conf | ||
| 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. | ||
| + | |||
| + | |||