Difference between revisions of "Telephone system:FreeSWITCH tricks"
m (added SRV info) |
(Elaborated a bit on the difference between '''sip_req_host''' and '''sip_to_host''') |
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| Line 5: | Line 5: | ||
= config snippets = | = config snippets = | ||
== dial by incoming domain == | == dial by incoming domain == | ||
| − | Typically, FreeSWITCH only looks at the destination number, so <code>1200@ackspace.nl</code> is treated the same as <code>1200@illegaleshow.nl</code>. | + | Typically, FreeSWITCH only looks at the destination number, so <code>1200@ackspace.nl</code> is treated the same as <code>1200@illegaleshow.nl</code> in the public dialplan. |
This snippet routes all calls destined to either the '''illegaleshow.nl''' domain or the '''illegaleshow''' extension to ''1200'' in its own dialplan. | This snippet routes all calls destined to either the '''illegaleshow.nl''' domain or the '''illegaleshow''' extension to ''1200'' in its own dialplan. | ||
| − | Note that there aren't many ''sip_to_host'' examples on the net (or the alternative ''sip_req_host'' for that matter) | + | Note that there aren't many ''sip_to_host'' examples on the net (or the less predictable alternative ''sip_req_host'' for that matter, also see [https://stackoverflow.com/questions/58947495/what-is-the-difference-between-the-invite-field-and-the-to-field-in-a-sip-invite this stackoverflow thread]): |
| + | * req or '''Request Uri''' is where the INVITE goes (he next hop address, a proxy if you will) | ||
| + | * '''To Uri''' is the actual destination to call | ||
<pre> | <pre> | ||
<extension name="Illegale show"> | <extension name="Illegale show"> | ||
Revision as of 21:30, 2 June 2021
This page provides a couple of tricks to ease working and debugging with FreeSWITCH
Contents
config snippets
dial by incoming domain
Typically, FreeSWITCH only looks at the destination number, so 1200@ackspace.nl is treated the same as 1200@illegaleshow.nl in the public dialplan.
This snippet routes all calls destined to either the illegaleshow.nl domain or the illegaleshow extension to 1200 in its own dialplan.
Note that there aren't many sip_to_host examples on the net (or the less predictable alternative sip_req_host for that matter, also see this stackoverflow thread):
- req or Request Uri is where the INVITE goes (he next hop address, a proxy if you will)
- To Uri is the actual destination to call
<extension name="Illegale show">
<condition regex="any">
<!-- sip_req_host -->
<regex field="${sip_to_host}" expression="illegaleshow.nl"/>
<regex field="destination_number" expression="^illegaleshow$"/>
<action application="set" data="domain_name=illegaleshow.nl"/>
<action application="transfer" data="1200 XML illegaleshow"/>
</condition>
</extension>
You only need to set some DNS SRV records and you're off to go:
_sip._tcp.illegaleshow.nl. 384 IN SRV 10 20 5060 sip.ackspace.nl. _sip._udp.illegaleshow.nl. 384 IN SRV 10 20 5060 sip.ackspace.nl.
Verify that they're active with: dig -t srv _sip._tcp.illegaleshow.nl
dial-string for Skinny and SIP
TODO: SCCP..
command line examples
You can run command line commands either as fs_cli -x "COMMAND" or by running the command line interface fs_cli and typing the COMMAND there
user_exists and user_data
Note that directory users have an id and optionally, a number-alias. Both can be used to register and check if it exists in the directory.
Typical for desk phones:
user_exists id 100 ackspace.nl- true
Typical for soft-phones (SIP client):
user_exists id ackspace ackspace.nl- true
Number from name or number
user_data 100@ackspace.nl attr number-aliasanduser_data ackspace@ackspace.nl attr number-alias- 100
Name from name or number
user_data 100@ackspace.nl attr idanduser_data ackspace@ackspace.nl attr id- ackspace
dial-string and sofia_contact
There is a user/ endpoint that uses the user's (or more regularly, the directory domain's) dial-string value to lookup the dial string address of the dialed user. This string usually contains the sofia_contact command:
sofia_contact ackspace@ackspace.nl- sofia/external/sip:ackspace@123.456.789.0:9876;transport=udp;user=phone;fs_nat=yes
Note: the contact URI only works on the registered user (use a number instead of a name if the user registered numerically; it happens, see Fritz!Box)
You can call a user by using: originate user/xopr@ackspace.nl 002 xml ackspace or originate user/196@ackspace.nl 002 xml ackspace.
