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guides:voip [2010/02/21 15:07] fishy ups, wrong header-levels. |
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> | + | |
- | 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? ===== | ===== Unable to upload MoH data? ===== | ||
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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?===== | ||
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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. | ||
+ | |||
+ |