FreePBX SIP Trunk

FreePBX SIP TrunkDocking

background: PBX1 is a virtual machine running FreePBX, whichnow needs to be connected via SIP TRUNK docking , PBX2, using PBX2 E1 The line calls out the phone.

PBX1 192.168.100.1

PBX2 192.168.100.2

PBX1on the configurationOneConfigurationTrunk

New SIP TRUNK

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TwoConfigurationOubtound Routers

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There are no matching rules, no matter what number you call, send the number directly to it's PBX2.

PBX2ConfigurationOneConfigurationSIP TRUNK

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TwoModifyAsteriskconfiguration file

Vi/etc/asterisk/extensions.conf

look for the from-trunk option, which is modified as follows:

[From-trunk]

Include = From-internal

Include = From-pstn

Note: Both sides PBX IP to allow each other!!! Also, if debug discovery is unsuccessful. First make sure that the phone number dialed on the PBX1 is successfully sent to PBX2 and the debug log is turned on. Then Add the From-trunk to the dial plan that normally uses the E1 line (and, of course, make sure the docking is ASTERISK is the dial plan that is used).

this way, The PBX1 can exhale the phone through the PBX. But the number of outgoing calls is not controllable. The outgoing display number is based on the match on the PBX2. For example, in an outbound route on my PBX2:

251 shows the 053181765959,

the prefix shown is 02131156123.

So, if I'm in PBX1 on the registered extension, when dialing, direct dial mobile phone number such as:1561XXXXX, will show 02131156123, and if I dial 251+ phone, will show 053181765959.

This article is from "Nothing Left" blog, please be sure to keep this source http://3357278.blog.51cto.com/3347278/1583135

FreePBX SIP Trunk